From rem-conf-request@es.net  Fri Oct  1 02:55:51 1999
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Date: Fri, 1 Oct 1999 01:23:17 -0600 (CST)
From: VELA GONZALEZ WENDY K <ad102336@mail.udlap.mx>
X-Sender: ad102336@mailweb
To: trouble@es.net, info@es.net, rem-conf@es.net, request-videophone@es.net
Subject: Could you do me a favor
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	Hi! I dont speak english very well, so I would like that you
help me to anwser a few questions, I am taking a curse o Marketing
by internet and I have to make a homework, I have to aswer these
questions about (Audio and video in Internet)
1.-History
2.-Definition
3.-How does them work  and
4.-The problem of (ANCHO DE BANDA...spanish)


I will be very happy if you could answering my questions...Thak you




From rem-conf-request@es.net  Fri Oct  1 03:00:06 1999
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Date: Fri, 1 Oct 1999 01:33:58 -0600 (CST)
From: VELA GONZALEZ WENDY K <ad102336@mail.udlap.mx>
To: VELA GONZALEZ WENDY K <ad102336@mail.udlap.mx>
cc: trouble@es.net, info@es.net, rem-conf@es.net, videophone@es.net
Subject: Re: Could you do me a favor
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On Fri, 1 Oct 1999, VELA GONZALEZ WENDY K wrote:

> 
> 
> 	Hi! I dont speak english very well, so I would like that you
> help me to anwser a few questions, I am taking a curse o Marketing
> by internet and I have to make a homework, I have to aswer these
> questions about (Audio and video in Internet)
> 1.-History
> 2.-Definition
> 3.-How does them work  and
> 4.-The problem of (ANCHO DE BANDA...spanish)
> 
> 
> I will be very happy if you could answering my questions...Thak you
> 
> 



From rem-conf-request@es.net  Fri Oct  1 03:05:25 1999
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Date: Fri, 1 Oct 1999 01:37:15 -0600 (CST)
From: VELA GONZALEZ WENDY K <ad102336@mail.udlap.mx>
To: trouble@es.net, info@es.net, rem-conf@es.net, videophone@es.net
Subject: Re: Could you do me a favor
In-Reply-To: <Pine.SUN.3.91.991001013341.18509B-100000@mailweb>
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On Fri, 1 Oct 1999, VELA GONZALEZ WENDY K wrote:

> 
> 
> On Fri, 1 Oct 1999, VELA GONZALEZ WENDY K wrote:
> 
> > 
> > 
> > 	Hi! I dont speak english very well, so I would like that you
> > help me to anwser a few questions, I am taking a curse o Marketing
> > by internet and I have to make a homework, I have to aswer these
> > questions about (Audio and video in Internet)
> > 1.-History
> > 2.-Definition
> > 3.-How does them work  and
> > 4.-The problem of (ANCHO DE BANDA...spanish)
> > 
> > 
> > I will be very happy if you could answering my questions...Thak you
> > 
> > 
> 



From rem-conf-request@es.net  Fri Oct  1 12:52:24 1999
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Message-ID: <014a01bf0c28$528d99a0$8a10740a@telefonica>
From: "Vicente Cid" <vicente.cid@telefonica.es>
To: <rem-conf@es.net>
References: <558.113143.511285@ns.bigbear.net>
Subject: remove vicente.cid@telefonica.es
Date: Fri, 1 Oct 1999 18:16:30 +0200
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remove vicente.cid@telefonica.es
_________________________________________________________________
Vicente Cid Cid
Telefónica de España
Gerente de Canales de Distribución Comercial
Territorio II (Galicia y Asturias)
C/Zamora 51 Bajo 36203 VIGO
Tel 986812730
Fax 986422607
vicente.cid@telefonica.es
----- Mensaje original -----
De: <mowhco@att.net>
Para: <rem-conf@es.net>
Enviado: sábado, 02 de octubre de 1999 3:59
Asunto: Homeworkers Needed!


> Dear Future Associate,
>
> You Can Work At Home & Set Your Own Hours.  Start earning Big
> Money in a short time
>
>                                     NO Newspaper Advertising!
>
> Your job will be to stuff and mail envelopes for our company. You
> will receive $.25 for each and every envelope you stuff and mail
> out.
>
> Just follow our simple instructions and you will be making money
> as easy as
> 1. 2. 3
>
> For example stuff and mail 200 envelopes and you will receive
> $50.00. Stuff and mail 1000 and you will receive $250.00. Stuff
> and mail 2000 and you will receive $500.00 and more
>
> Never before has there been an easier way to make money from
> home!
>
> Our Company's Home Mailing Program is designed for people with
> little or no experience and provides simple, step by step
> instructions.
>
> There is no prior experience or special skills necessary on your
> part, Just stuffing envelopes.
>
> We need the help of honest and reliable home workers like you.
> Because we are overloaded with work and have more than our staff
> can handle. We have now expanded our mailing program and are
> expecting to reach millions more with our offers throughout the
> US and Canada.
>
> Our system of stuffing and mailing envelopes is very simple and
> easy to do!
> You will not be required to buy envelopes or postage stamps.
>
> We will gladly furnish all circulars at no cost to you. We assure
> you that as a participant in our program you will never have to
> mail anything objective or offensive.
>
> There are no quotas to meet, and there no contracts to sign. You
> can work as much, or as little as you want. Payment for each
> envelope you send out is Guaranteed!
>
> Here is what you will receive when you get your first Package.
> Inside you will find 100 envelopes, 100 labels and 100 sales
> letters ready to stuff and mail
>
> As soon as you are done with stuffing and mailing these first
> letters, your payment will arrive shortly, thereafter. All you
> have to do is to order more free supplies and stuff and mail more
> envelopes to make more money.
>
> Our sales literature which you will be stuffing and mailing will
> contain
> information outlining our highly informative manuals that we are
> advertising nationwide.  As a free gift you will receive a
> special manual valued at  $24.95, absolutely free, just for
> joining our Home Mailers Program.
>
> Plus you will get your own special code number, so that we will
> know how much you are to get paid.  And to make re-ordering of
> more envelopes, that our company supplies very simple for you.
>
> We are giving you this free bonus because we want you to be
> confident in our company and to ensure that we will be doing
> business with you for a long time.
>
> Benefits Of This Job:
>
> 1. You do not have to quit your present job, to earn more money
> at home
> 2. You can make between $2,500 to $4,500 a month depending on the
> amount of time you are willing to spend stuffing and mailing
> envelopes
> 3. This is a great opportunity for the students, mothers,
> disabled persons or those who are home bodies.
>
> To secure your position and to show us that you are serious about
> earning extra income at home we require a one-time registration
> fee of $35.00.
> This fee covers the cost of your initial start up package,  which
> includes 100 envelopes, 100 labels and 100 sales letters and a
> manual, your registration fee will be refunded back to you
> shortly thereafter.
>
> Money Back Guarantee!
>
> We guarantee that as soon as you stuff and mail your first 300
> envelopes You will be paid $75.00 and your registration fee will
> be refunded.
>
> Many of you wonder why it is necessary to pay a deposit to get a
> job. It is because we are looking for people that seriously want
> to work from home.
>
> *  If 3.000 people told us they wanted to start working from home
> and we sent out 3.000 packages free to every one.  And then half
> of the people decided not to work, this would be a potential loss
> of more than $60,000 in supply's and shipping that we have sent
> out to people that don't want to work
>
> We have instituted this policy to make sure that you really want
> to work and at least finish your first package.
>
> To Get Started Today Please Enclose Your Registration Fee of $35
> Check,Cash Or Money Order and fill out the application below and
> mail to:
>
> MOHW Co
> PMB
> 11054 Ventura Blvd #126
> Studio City, CA 91604
>
> Name_____________________________________________________
>
> Address___________________________________________________
>
> City____________________________________ State______________
>
> Zip Code________________
>
> Telephone Number(s)_________________________________________
>
> E-mail Address______________________________________________
>
>
>
> For all orders, please allow seven (7) days for delivery and up
> to 10 days. Cash and Money Orders will result in faster shipping of your
> package.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>




From rem-conf-request@es.net  Fri Oct  1 13:55:28 1999
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From: "Vicente Cid" <vicente.cid@telefonica.es>
To: <freetravel@boom.com>, <rem-conf@es.net>
References: <199910010151.DAA06569@micro.iternet.it>
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remove vicente.cid@telefonica.es
_________________________________________________________________
Vicente Cid Cid
Telefónica de España
Gerente de Canales de Distribución Comercial
Territorio II (Galicia y Asturias)
C/Zamora 51 Bajo 36203 VIGO
Tel 986812730
Fax 986422607
vicente.cid@telefonica.es
----- Mensaje original -----
De: <freetravel@boom.com>
Para: <rem-conf@es.net>
Enviado: lunes, 15 de marzo de 1999 1:01
Asunto: Free Vacation when you Request the Info


>           GET YOUR FREE VACATION VOUCHER WHEN
>                 VISITING OUR WEB SITE!
>
>               ONLY 14 ASSOCIATES NEEDED!
>   We will train hard working and ethical candidates.
>
>            HUGE COMPENSATION PLAN AVAILABLE!
> This is your chance to get in on the ground floor of the
>   Multi-Billion-Dollar Home Based Business Revolution.
>
>       Receive a COMPLIMENTARY Weekend Get Away when
>                    visiting our site.
>
>
>      http://3626046468/ut/travelfree/index2.html
>
> <a href="http://3626046468/ut/travelfree/index2.html">Click Here
> FOR FREE INFORMATION</a>
>
>




From rem-conf-request@es.net  Fri Oct  1 23:24:01 1999
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Date: Fri, 01 Oct 1999 23:01:04 -0400
To: rem-conf@es.net
From: Michael Aldrich <maldrich@ne.mediaone.net>
Subject: Send more info
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--=====================_47387804==_.ALT
Content-Type: text/plain; charset="us-ascii"; format=flowed



At 08:13 PM 9/17/99 -0700, you wrote:

>Take a look at this......
>
>Greetings....
>
>THE NEWS THAT HAS BEEN RUMORED FOR WEEKS
>
>IS HERE
>
>NEWS THAT WILL TAKE THE WORLD BY STORM!
>
>REGISTER NOW TO SECURE BEST POSITION DURING PRE-LAUNCH
>
>Just imagine. Thousands upon Thousands of Premium Products
>
>and Services all categorized for your easy use
>
>THE LARGEST DIRECTORY OF FREE OFFERS IN THE WORLD!
>
>To Start With . Take the tour and you can have your very
>
>own
>
>Digital Satellite System Certificate, valued at
>
>$149 FREE
>
>We know that no one has to be a computer genius or know everything about
>
>the Internet to profit from it .Combine FREE with the dynamics of
>
>cyber space and you have the World's easiest and most lucrative business
>
>Work will simply be Sharing FREE goods and services with others
>
>Take this for starters .. All subscribers will have FREE
>
>INTERNET
>
>and E-MAIL ACCESS That is just the beginning!
>
>But .An immediate saving on your current overheads.
>
>YOU WILL ALSO HAVE .. FREE
>
>* Software allowing long distance calls at zero dollars per minute.
>
>* $100 airline check
>
>* Encarta Encyclopedia 99
>
>* Personal Computer
>
>* Las Vegas vacation
>
>* Thousands upon Thousands of Free Products and Services
>
>Just click your reply button now and type the words "send more info " in 
>your subject box and I will send you the website where you can get the 
>rest of the information and join straightaway.
>
>===========================================================================
>
>&gt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt 
>&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt 
>&lt&lt&lt&lt&lt&lt&lt&lt
>
>&gtPLEASE NOTE: You have received this email because your email address is
>
>&gtpart of our "in house" list. You or someone you know has sent your
>
>&gtemail address to us in the past. Our team promotes professional and
>
>&gtresponsible use of direct email marketing. If this message is in
>
>&gterror we apologize. Further transmissions to you by the sender of this
>
>&gtemail may be stopped at no cost to you by sending a reply to this
>
>&gtemail address with the word "remove" in the subject line.
>
>&gt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt 
>&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt&lt 
>&lt&lt&lt&lt&lt&lt&lt&lt
>
>=======================================

*******************************************************************************
Michael Aldrich                                      White Pine Software
Sr. Support Technician                              542 Amherst St.
Phone: (603) 886-9050                              Nashua NH. 03063
Fax: (603) 886-9051                                  http://www.wpine.com

Please visit our website for all the latest upgrade's and
enhancement's to White Pine Software's Awesome Product's.
**************************************************************************** 
***  
--=====================_47387804==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<br>
<br>
At 08:13 PM 9/17/99 -0700, you wrote:<br>
<br>
<font size=2 color="#0000FF"><b><blockquote type=cite cite>Take a look at
this......</b> <br>
<br>
</font><font size=2 color="#FF0000"><b>Greetings....<br>
</b><br>
</font><font size=2 color="#0000FF">THE NEWS THAT HAS BEEN RUMORED FOR
WEEKS</b> <br>
<br>
<b>IS HERE</b> <br>
<br>
<b>NEWS THAT WILL TAKE THE WORLD BY STORM!</b> <br>
<br>
<b>REGISTER NOW TO SECURE BEST POSITION DURING PRE-LAUNCH<br>
</b><br>
Just imagine. Thousands upon Thousands of Premium Products</b> <br>
<br>
<b>and Services all categorized for your easy use</b> <br>
<br>
</font><font size=2 color="#FF0000"><b>THE LARGEST DIRECTORY OF FREE
OFFERS IN THE WORLD!<br>
</b><br>
To Start With . Take the tour and you can have your very</b> <br>
<br>
<b>own</b> <br>
<br>
<b>Digital Satellite System Certificate, valued at</b> <br>
<br>
<b>$149 FREE</b> <br>
<br>
<b>We know that no one has to be a computer genius or know everything
about</b> <br>
<br>
<b>the Internet to profit from it .Combine FREE with the dynamics of</b>
<br>
<br>
<b>cyber space and you have the World's easiest and most lucrative business</b> <br>
<br>
<b>Work will simply be Sharing FREE goods and services with others</b> <br>
<br>
<b>Take this for starters .. All subscribers will have FREE</b> <br>
<br>
<b>INTERNET</b> <br>
<br>
<b>and E-MAIL ACCESS That is just the beginning!</b> <br>
<br>
<b>But .An immediate saving on your current overheads.</b> <br>
<br>
<b>YOU WILL ALSO HAVE .. FREE</b> <br>
<br>
<b>* Software allowing long distance calls at zero dollars per minute.</b> <br>
<br>
<b>* $100 airline check</b> <br>
<br>
<b>* Encarta Encyclopedia 99</b> <br>
<br>
<b>* Personal Computer</b> <br>
<br>
<b>* Las Vegas vacation</b> <br>
<br>
<b>* Thousands upon Thousands of Free Products and Services</b> <br>
<br>
<b>Just click your reply button now and type the words &quot;send more info &quot; in your subject box and I will send you the website where you can get the rest of the information and join straightaway.</b> <br>
<br>
<b>===========================================================================</b> <br>
<br>
<b>&amp;gt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt&amp;lt</b> <br>
<br>
<b>&amp;gtPLEASE NOTE: You have received this email because your email address is</b> <br>
<br>
<b>&amp;gtpart of our &quot;in house&quot; list. You or someone you know has sent your</b> <br>
<br>
<b>&amp;gtemail address to us in the past. Our team promotes professional and</b> <br>
<br>
<b>&amp;gtresponsible use of direct email marketing. If this message is in</b> <br>
<br>
<b>&amp;gterror we apologize. Further transmissions to you by the sender of this</b> <br>
<br>
<b>&amp;gtemail may be stopped at no cost to you by sending a reply to this</b> <br>
<br>
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<br>
<b>======================================= </font></b></blockquote><br>

*******************************************************************************<br>
<b>Michael Aldrich</b>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <b>White Pine Software<br>
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Fax: (603) 886-9051&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href="http://www.wpine.com/" eudora="autourl"><font color="#0000FF"><u>http://www.wpine.com<br>
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enhancement's to White Pine Software's Awesome Product's. <br>
******************************************************************************* </font></html>

--=====================_47387804==_.ALT--




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From rem-conf-request@es.net  Sun Oct  3 02:10:58 1999
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Best Regards,
Cissy

----- "Live is about turning the things you want to do,
                                 to the things that you've done." ------

 ***********************************************************************
 * Cissy Fang Liu                                          EEB418      *
 * Ph.D Student                                  Tel: (213)740-4657(O) *
 * Signal and Image Processing Institute              		       *
 * Department of Electrical Engineering          Email: fanliu@usc.edu *
 * University of Southern California        http://biron.usc.edu/~fliu *
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From rem-conf-request@es.net  Sun Oct  3 12:34:59 1999
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To: rem-conf@es.net
Subject: WG last call: MIB and tones drafts
Date: Sun, 03 Oct 1999 17:12:11 +0100
Message-ID: <4175.938967131@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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To the AVT working group,

This is to announce a two week working group last call on the following
drafts:

Title     : RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
Author(s) : H. Schulzrinne, S. Petrack
Filename  : draft-ietf-avt-tones-01.txt
Pages     : 25
Date      : 27-Sep-99

Title     : Real-Time Transport Protocol Management Information Base
Author(s) : M. Baugher, I.  Suconick,  B. Strahm
Filename  : draft-ietf-avt-rtp-mib-06.txt
Pages     : 37
Date      : 28-Sep-99

The DTMF draft addresses the concerns raised during the Oslo meeting, and
the MIB addresses comments raised during the previous working group last
call held earlier this year.

Please send comments to the rem-conf@es.net list before 18th October. If
there are no serious objections, we intend to request the IESG to consider
these for publication as proposed standards at that time.

Colin



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Subject: Search Engine Registration    adv
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I saw your listing on the internet.  I work
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Remove



From rem-conf-request@es.net  Tue Oct  5 02:23:40 1999
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Date: Tue, 5 Oct 1999 01:46:36
From: <travellb@boom.com>
Subject: Promote Travel Industry From Home.
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From rem-conf-request@es.net  Tue Oct  5 08:15:08 1999
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to: rem-conf@es.net
Subject: UCL MBone Announcement: Seminar on IP Evolution by Steve Deering
Date: Tue, 05 Oct 1999 08:21:13 +0100
Message-ID: <420.939108073@cs.ucl.ac.uk>
From: Jon Crowcroft <J.Crowcroft@cs.ucl.ac.uk>
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we'll be mbone-ing a seminar this friday at 1pm BST (see sdr for the
Hourglass session), and 
http://www.cs.ucl.ac.uk/dept-seminar-abstracts/dept-seminar-991008.html
for abstract

its only on a best effort basis, so if things go horribly wrong, we
are not going to chase any routing/congestion problems, but hopefully
you may be able to see some of it....

thanks
jon



From rem-conf-request@es.net  Tue Oct  5 13:06:02 1999
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Reply-To: "M. Reha Civanlar" <civanlar@research.att.com>
From: "M. Reha Civanlar" <civanlar@research.att.com>
To: <rem-conf@es.net>
Subject: ID on pointer payload format
Date: Tue, 5 Oct 1999 12:26:36 -0400
Organization: AT&T Labs - Research
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I have just posted an updated ID describing the RTP payload
format for transmitting real-time pointers. If you want to
take
a look at it before it becomes available on the regular
web site, it is available at:

http://www.research.att.com/~mrc/draft-ietf-avt-pointer-01.t
xt
or
http://www.research.att.com/~mrc/draft-ietf-avt-pointer-01.p
s

This update incorporates the modifications/additions as
decided in the Oslo meeting. As always, comments are
welcome.

Best regards.

------------------------------------------------------------
-----------
M. Reha Civanlar
Division Manager, AT&T Labs - Research
100 Schultz Drive, Red Bank, NJ 07701, U.S.A
civanlar@research.att.com
Phone: +1 732 345 3305 Fax:+1 732 345 3033
http://www.research.att.com/info/mrc





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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 10/6 Resource Management in Operating Systems --Timothy
  Roscoe, Applied Technology Lab Sprint Communications 
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Berkeley Multimedia, Graphics and Interfaces Seminar

Resource Management in Operating Systems

Wednesday October 6, 1999 
1:00-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 


Timothy Roscoe
Applied Technology Lab Sprint Communications

This is a talk about how multimedia research and operating systems
interact, in a few cases. Traditionally, multimedia applications research
in academia and quite a few corporate labs has been based on dedicated
hardware of one sort or another, either built specially for the application
or else a general-purpose computer given over entirely to one processing
job. The problem is usually not that the hardware lacks the resources, but
that the system is incapable of sharing the resources in the appropriate
way, including dynamically reallocating resources and policing resource
usage. 

A few operating systems have appeared recently which directly address these
problems, and in doing so require multimedia application designers to
reorient their thinking away from virtual, abstracted resources to
physical, controlled resources. This doesn't necessarily make writing
programs harder, and can actually result in much simpler and more robust
multimedia applications. 

I'll talk about this different way of thinking about systems, especially in
relation to the Nemesis operating system.
---------------

The seminar is broadcast live on the Internet Mbone and as a Real
Networks G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player:

rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
   http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving
the announcement you may be able to receive the session by manually
configuring the client programs ('vic', and 'vat') with the session
addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to
replays of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 

____________________________________________

Florissa Colina
Berkeley Multimedia Research Center (BMRC)
http://bmrc.berkeley.edu/
626 Soda Hall #1776
University of California
Berkeley, CA 94720-1776
Phone: (510) 643-0800   FAX: (510) 642-5615  
Email: florissac@bmrc.berkeley.edu



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From: <easenterprises@hotmail.com>
To: <rem-conf@es.net>
Date: Tue, 5 Oct 1999 15:19:57
Message-Id: <773.856522.812828@mailer100.gte.net>
Subject: Quick Cash From Home $$$
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>Subject: Quick Cash From Home $$$
>Date: Mon, 4 Oct 1999 16:32:10
>
> >>>>>>>>>HERE IS THE INFO YOU REQUESTED, THIS IS NOT SPAM, YOU 
ARE
>RECEIVING THIS E-MAIL BECAUSE YOU OR SOMEONE ELSE REQUESTED THIS
>INFORMATION. IF YOU WISH TO BE PERMANENTLY REMOVED FROM OUR
>MAILING LIST. PLEASE SEND AN E-MAIL WITH THE WORD "REMOVE" IN THE
>SUBJECT LINE TO:
>easenterprises@hotmail.com. OUR COMPANY DOES NOT SUPPORT THE
>SENDING OF UNSOLICITED E-MAILS.
> >>>>>>>>>*******************************************************
> >>>>>>>>>>>>>>
> >>>>>>>>>      WORK AT HOME USING YOUR COMPUTER!!!
>
> >>>>>>>>>*******************************************************
> >>>>>>>>>
>Dear Friend,
> >>>>>>>>>
>You can earn $46,000 or more in next the 90 days sending e-mail,
>seem impossible?  Read on for details (no, there is no 
'catch')...
> >>>>>>>>>
> >>>>>>>>>"AS SEEN ON NATIONAL T.V."
> >>>>>>>>>
> >>>>>>>>>Thank you for your time and Interest.
> >>>>>>>>>This is the letter you've been reading about in the 
news
>lately.
> >>>>>>>>>
>Due to the popularity of this letter on the internet,
>a major nightly news program recently devoted an entire show to
>the  investigation of the program described below to see, if it
>really can make people money.
> >>>>>>>>>
>The show also investigated whether or not the program was legal.
>Their findings proved once and for all that there are,
>absolutely no laws prohibiting the participation in the program.
>This has helped to show people that this is a simple,
>harmless and fun way to make some extra money at home.
>The results of this show has been truly remarkable.
>So many people are participating that those involved are doing,
>much better than ever before.  Since everyone makes more as
>more people try it out, its been very exciting to be a part of
>lately.
> >>>>>>>>>You will understand once you experience it.
> >>>>>>>>>
> >>>>>>>>>"HERE IT IS BELOW"
> >>>>>>>>>
> >>>>>>>>>================================================
> >>>>>>>>>================================================
> >>>>>>>>>
> >>>>>>>>>              *** Print This Now For Future Reference 
***
> >>>>>>>>>
> >>>>>>>>>The following income opportunity is one you may be
>interested in taking a look at.  It can be started with VERY
>LITTLE investment and the income return is TREMENDOUS!!!
> >>>>>>>>>
> >>>>>>>>>$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$
> >>>>>>>>>
> >>>>>>>>>If you would like  to make at least $46,000 in less 
than
>90 days!
> >>>>>>>>>Please read the enclosed program...THEN READ IT 
AGAIN!!!
> >>>>>>>>>
> >>>>>>>>>$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$
> >>>>>>>>>
>THIS IS A LEGITIMATE, LEGAL, MONEY MAKING OPPORTUNITY.   It does
>not require you to come into contact with people, do any hard
>work, and best of all, you never have to leave
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>you'll get that big break that you've been waiting for, THIS IS
>IT!  Simply follow the instructions, and your dreams will come
>true.
> >>>>>>>>>This multi-level e-mail order marketing program works
> >perfectly...100% EVERY TIME.  E-mail is the sales tool of the
>future.  Take advantage of this non-commercialized method of
>advertising NOW!!! The longer you wait, the more people will be
>doing business using e-mail.  Get your piece of this action!!!
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>MULTI-LEVEL MARKETING (MLM) has finally gained
>respectability.  It is being taught in the Harvard Business
>School, and both Stanford Research and
>the Wall Street Journal have stated that between 50% and 65% of
>all goods and services will be sold through multi-level methods 
by
>the mid to late 1990's.  This is a Multi-Billion Dollar industry
>and of the 3,500,000 millionaires in the WORLD, 20% ( 700,000 )
>made their fortune in the last several
>years in MLM.  Moreover, statistics show that over 100 people
>become millionaires everyday through
>Multi-Level Marketing.
>You may have heard this story before, but over the summer Donald
>Trump (A MULTI-BILLIONAIRE, ONE OF THE WEALTHIEST MEN IN THE
>WORLD)
>made an appearance on the David Letterman show. Dave
>asked him what he would do if he lost everything and had to start
>over from scratch. Without hesitating, Trump said he would find a
>good network marketing
>company and
>get to work. The audience started to hoot and boo him. He looked
>out at the audience and dead-panned his
>response  "That's why I'm sitting up here and you are all sitting
>out there!"
> >>>>>>>>>
> >>>>>>>>>With network marketing you have two sources of income.
>Direct commissions from sales you make yourself and commissions
>from sales made by people you introduce to the business.
>Residual income is the secret of the wealthy. It means investing
>time or money once and getting paid again and again and again.
>In network marketing, it also means getting paid for the work of
>others. This program is currently being utilized in more than 50
>different countries across the world.
>
>The enclosed information is something I almost let slip through 
my
>fingers. Fortunately, sometime later I re-read everything and 
gave
>some thought and studyto it.
> >>>>>>>>>
> >>>>>>>>>My name is Johnathon Rourke.  Two years ago, the
>corporation I
>worked at for the past twelve years down-sized and my position 
was
>eliminated.  After unproductive job interviews, I decided to open
>my own business.
>Over the
>past year, I incurred
>many unforeseen financial problems.  I owed my family, friends 
and
>creditors over $35,000.  The
>economy was taking a toll on my business and I just couldn't seem
>to make ends meet.  I had to refinance and borrow against my home
>to support my family and struggling business.  AT THAT MOMENT
>something significant happened in my life and I am writing to
>share the experience in hopes that this will change your life
>FOREVER FINANCIALLY!!!
> >>>>>>>>>
> >>>>>>>>>In mid December, I received this program via e-mail.  
Six
>month's prior to receiving this program I had been sending away
>for information on various business opportunities.  All of the
>programs I received, in my opinion, were not cost effective.  
They
>were either too difficult for me to comprehend or the initial
>investment was too much for me to risk to see if they would work
>or not. One claimed that I would make a million dollars in one
>year...it didn't tell me I'd have to write a book to make it!
> >>>>>>>>>
>But like I was saying, in December of 1997 I received this
>program.I didn't send for it, or ask for
>it, they just got my name off a mailing list. THANK GOODNESS FOR
>THAT!!! After reading it
>several times, to make sure I was reading it correctly, I 
couldn't
>believe my eyes.  Here was a MONEY MAKING PHENOMENON. I could
>invest as much as I wanted to start, without putting me further
>into debt.  After I got a pencil and paper and figured it out, I
>would at least get my money back.  But like most of you I was
>still a little skeptical and a little worried about the legal
>aspects of it all. So I checked it out
>with the U.S. Post Office (1-800-725-2161 24-hrs) and they
>confirmed that it is indeed legal! After determining the program
>was LEGAL and NOT A CHAIN LETTER, I decided "WHY NOT."
> >>>>>>>>>
>Initially I sent out 100,000 e-mails.  It cost me about $15 for 
my
>time on-line.  The great thing about e-mail is that I don't need
>any money for printing to send out the program, and because all 
of
>my orders are fulfilled via e-mail,the only expense is my time.  
I
>am telling you like it is, I hope it doesn't turn you off, but I
>promised myself that I would not "rip-off" anyone, no matter how
>much money it cost me.
> >>>>>>>>>
> >>>>>>>>>In less than one week, I was starting to receive orders
>for REPORT #1.  By January 13, I had received 26 orders for 
REPORT
>#1. Your goal is to "RECEIVE at least 20 ORDERS FOR REPORT #1
>WITHIN 2 WEEKS.  IF YOU DON'T, SEND OUT MORE PROGRAMS UNTIL YOU
>DO!"  My first step in making $46,000 in 90 days was done. By
>January 30, I had received 196 orders for REPORT #2.  Your goal 
is
>to "RECEIVE AT LEAST 100+ ORDERS FOR REPORT #2
>WITHIN 2 WEEKS. IF NOT, SEND OUT MORE PROGRAMS UNTIL YOU
>DO. ONCE YOU HAVE 100 ORDERS, THE REST IS EASY, RELAX,
> >>>>>>>>>
> >>>>>>>>>YOU WILL MAKE YOUR $46,000 GOAL."  Well, I had 196 
orders
>for
> >>>>>>>>>REPORT #2, 96 more than I needed.  So I sat back and
>relaxed.
> >>>>>>>>>By March 1, of my e-mailing of 100,000, I received
>$58,000 with more coming in every day.
> >>>>>>>>>
> >>>>>>>>>I paid off ALL my debts and bought a much needed new 
car.
> >>>>>>>>>Please take time to read the attached program, IT WILL
>CHANGE YOUR LIFE FOREVER!!!
> >>>>>>>>>Remember, it won't work if you don't try it.  This
>program does work, but you must follow it
>EXACTLY!  Especially the rules of not trying to place your name 
in
>a different place. It won't
>work, you'll lose out on a lot of money!   In order for this
>program to work, you must meet your goal of 20+ orders for REPORT
>#1, and 100+ orders for REPORT #2 and you will make $46,000 or
>more in 90 days. I AM LIVING PROOF THAT IT WORKS!!!
>If you choose not to participate in this program, I am sorry.
>Itreally is
>a great opportunity with little cost or risk to you.  If you
>choose to participate, follow the program and you will be on your
>way to financial security.If you are a fellow business owner and
>are if financial trouble like I was,or you want to start your own
>business, consider this a sign. I DID!
> >>>>>>>>>
> >>>>>>>>>                       Sincerely,
> >>>>>>>>>
> >>>>>>>>>                      Johnathon Rourke
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>P.S. Do you have any idea what 11,700  $5 bills 
($58,000)
>look like piled up on a kitchen table!!IT'S AWESOME!
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>A PERSONAL NOTE FROM THE ORIGINATOR OF THIS
> >>>>>>>>>PROGRAM:
> >>>>>>>>>
>By the time you have read the enclosed program and reports, you
>should have concluded that such a program, and one that is legal,
>could not have been created by an amateur.
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>Let me tell you a little about myself.  I had a
>profitable business for 10 years.  Then in 1979 my
>business began falling off.  I was doing the same things that 
were
>previously successful for me,
>but it wasn't working.  Finally, I figured it out.  It wasn't me,
>it was the economy.  Inflation and recession had replaced the
>stable economy that had been with us since 1945. I don't have to
>tell you what happened to the unemployment rate... because many 
of
>you know from first hand
>experience.  There were more failures and bankruptcies than ever
>before. The middle class was vanishing.  Those who knew what they
>were doing invested wisely and moved up.  Those who did not,
>including those who never had anything to save or invest, were
>moving down into the ranks of the poor.  As the saying goes, "THE
>RICH GET RICHER AND THE POOR GET POORER."  The traditional 
methods
>of making money will never allow you to "move up" or "get rich",
>inflation will see to that.
> >>>>>>>>>
> >>>>>>>>>You have just received information that can give you
>financial freedom for the rest of your life, with "NO RISK" and
>"JUST A LITTLE BIT OF EFFORT."  You can make more money in the
>next few months than you have ever imagined.
> >>>>>>>>>
> >>>>>>>>>I should also point out that I will not see a penny of
>this money, nor anyone else who has provided a testimonial for
>this program. I have already made over 4 MILLION DOLLARS!  I have
>retired from the program after sending out over 1,600,000
>programs. Now I have several offices that make this and several
>other programs here and overseas.
> >>>>>>>>>
> >>>>>>>>>Follow the program EXACTLY AS INSTRUCTED.  Do not 
change
>it in any way.  It works exceedingly well as it is now.  Remember
>to e-mail a copy of this exciting
>report to everyone you can think of. One of the people you send
>this to may send out 100,000 or more...and your name will be on
>everyone of them!  Remember though, the more you send out the 
more
>potential customers you will reach.
> >>>>>>>>>
> >>>>>>>>>So my friend, I have given you the ideas, information,
>materials and opportunity to become financially independent, IT 
IS
>UP TO YOU NOW!
> >>>>>>>>>
> >>>>>>>>>"THINK ABOUT IT" Before you delete this program from 
your
>mailbox, as I almost did, take a little time to read it and 
REALLY
>THINK ABOUT IT.  Get a pencil and figure out what could happen
>when YOU participate.
>Figure out the worst possible response and no matter how you
>calculate it, you will still make a lot of money!  You will
>definitely get back what you invested.  Any doubts you have will
>vanish when your first orders come in.  IT WORKS!
> >>>>>>>>>                                      Jody Jacobs,
>Richmond, VA
> >>>>>>>>>
> >>>>>>>>>HERE'S HOW THIS AMAZING PROGRAM WILL MAKE YOU THOUSANDS
>OF DOLLAR$
> >>>>>>>>>
> >>>>>>>>>INSTRUCTIONS:
> >>>>>>>>>This method of raising capital  REALLY WORKS 100% EVERY
>TIME. I am sure that you could use up to $46,000 or more in the
>next 90 days. Before you say "BULL... ", please read this program
>carefully.
> >>>>>>>>>This is not a chain letter, but a perfectly legal money
>making opportunity.  Basically, this is what you do:  As with all
>multi-level businesses, we build our business by recruiting new
>partners and selling our products.  Because of the global nature
>of the internet,
>you will be able to recruit new multi-level business partners 
from
>all over the world, and we offer a product for EVERY dollar sent.
>  YOUR ORDERS COME BY MAIL AND ARE FILLED BY E-MAIL, so you are 
not
>involved in personal selling.  You do it privately in your own
>home, store or office.  This is the GREATEST Multi-Level Mail
>Order
> >>>>>>>>>Marketing anywhere:
> >>>>>>>>>
> >>>>>>>>>This is what you MUST do:
> >>>>>>>>>
> >>>>>>>>>1. Order all 5 reports shown on the list below (you 
can't
>             sell them if you don't order them).
> >>>>>>>>>
> >>>>>>>>>*  For each report, send $5.00 CASH, the NAME & NUMBER
> >>>>>>>>>OF THE REPORT YOU ARE ORDERING, YOUR E-MAIL ADDRESS, 
and
> >>>>>>>>>YOUR NAME & RETURN ADDRESS (in case of a problem) to 
the
>          person whose name appears on the list next to the 
report.
> >>>>>>>>>MAKE SURE YOUR RETURN ADDRESS IS ON YOUR ENVELOPE
> >>>>>>>>>IN CASE OF ANY MAIL PROBLEMS!
> >>>>>>>>>
> >>>>>>>>>      *  When you place your order, make sure you order
>each of the five reports.  You will need all five reports so that
>you can save them on your computer and resell them.
> >>>>>>>>>
> >>>>>>>>>      *  Within a few days you will receive, via 
e-mail,
>each of the five reports. Save them on your computer so they will
>be accessible for you to send to the 1,000's of people who will
>order them from you.
> >>>>>>>>>
> >>>>>>>>>2.  IMPORTANT-- DO NOT alter the names of the people 
who
>are listed next to each report, or their sequence on the list, in
>any way other than is instructed below in steps "a"
>through "g" or you will lose out on the majority of your profits.
>  Once you understand the way this works, you'll also see how it
>doesn't work if you change it. Remember, this method has been
>tested, and if you alter it, it will not work.
> >>>>>>>>>
> >>>>>>>>>a.  Look below for the listing of available reports.
> >>>>>>>>>
> >>>>>>>>>b.  After you've ordered the five reports, take this
> >>>>>>>>>         advertisement and remove the name and address
>under
> >>>>>>>>>         REPORT #5. This person has made it through the
>cycle and is no doubt counting their $46,000!  Also, change the
>name of the company, the address, and the remove e-mail address 
on
>the top of this document to your own.
> >>>>>>>>>
> >>>>>>>> c.  Move the name and address under REPORT #4 down to
> >>>>>>>>>         REPORT #5.
> >>>>>>>>>
> >>>>>>>>>d.  Move the name and address under REPORT #3 down to
> >>>>>>>>>         REPORT #4.
> >>>>>>>>>
> >>>>>>>>>e.  Move the name and address under REPORT #2 down to
> >>>>>>>>>         REPORT #3.
> >>>>>>>>>
> >>>>>>>>>f.  Move the name and address under REPORT #1 down to
> >>>>>>>>>          REPORT #2.
> >>>>>>>>>
> >>>>>>>>>g.  Insert your name/address in the REPORT #1 position.
> >>>>>>>>>
> >>>>>>>>>Please make sure you copy every name and address
> >>>>>>>>>ACCURATELY!
> >>>>>>>>>
> >>>>>>>>>3.  Take this entire letter, including the modified 
list
>of names, and save it to your computer.  Make NO changes to the
>instruction portion of this letter.
> >>>>>>>>>
> >>>>>>>>>Your cost to participate in this is practically nothing
>(surely you can afford $25). You obviously already have an
>Internet connection and e-mail is FREE!
> >>>>>>>>>
> >>>>>>>>>To assist you with marketing your business on the
>internet, the 5 reports you purchase will provide you with
>invaluable marketing information which includes how to send bulk
>e-mails, where to find thousands of free classified ads and much,
>much more.
> >>>>>>>>>
> >>>>>>>>>In addition you will be provided with information on
>Internet Marketing Clubs such as INTERNET MARKETING
>RESOURCES(IMR): This is one the premiere  internet marketing 
clubs
>on the INTERNET.  This club provides a forum where interne
>marketers from all over the world can exchange ideas and secrets
>on Internet Marketing.  In addition, members of this club are
>provided free internet marketing tools and services for the
>Do-Yourself-Internet-Marketer.  They will provide you with free
>bulk e-mail software and up to 1,000,000 fresh e-mail addresses
>each week.   This club will provide you with hundreds of free
>resources which include:
>How to obtain free web sites,   how to obtain top rankings in
>search engines for your web-site, how to send bulk e-mail into
>AOL and Compuserve, how to market your products on
>newsgroups, free classified ads, electronic malls, bulletin
>boards, banner ads and much more.
> >>>>>>>>>
> >>>>>>>>>There are two primary methods of building your 
downline:
> >>>>>>>>>
> >>>>>>>>>METHOD #1: SENDING BULK E-MAIL
> >>>>>>>>>
>Let's say that you decide to start small, just to see how it 
goes,
>and we'll assume you and all those involved send out only 2,000
>programs each.  Let's also assume that the mailing receives a 
0.3%
>response.  Using a good list the response could be much better.
>Also, many people will send out hundreds of thousands of programs
>instead of 2,000.  But continuing with this example, you send out
>only 2,000 programs. With a 0.3% response, that is only 6 orders
>for REPORT #1.  Those 6 people respond by sending out 2,000
>programs each for a total of 12,000.  Out of those 0.3%, 36 
people
>respond and order
>REPORT #2.  Those 36 mail out 2,000 programs each for a total of
>72,000. The 0.3% response to that is 216 orders for REPORT #3.
>Those 216 send out 2,000 programs each for a 432,000 total.  The
>0.3% response to that is 1,296 orders for REPORT #4. Those 1,296
>send out 2,000 programs each for a 2,592,000 total.  The 0.3%
>response to that is 7,776 orders for REPORT #5. That's 7,776 $5
>bills for you. CASH!!! Your total income in this example is $30 +
>$180 + $1,080 + $6,480 + $38,880 for a total of $46,650!!!
> >>>>>>>>>
> >>>>>>>>>REMEMBER FRIEND, THIS IS ASSUMING 1,994 OUT OF THE 
2,000
>PEOPLE YOU MAIL TO WILL DO ABSOLUTELY NOTHING AND TRASH THIS
>PROGRAM!  DARE TO THINK FOR A MOMENT WHAT WOULD HAPPEN IF
>EVERYONE,
>OR HALF SENT OUT 100,000 PROGRAMS INSTEAD OF 2,000.
>Believe me, many people will do just that, and more!  By the way,
>your cost to participate in this is practically nothing.  You
>obviously already have an internet connection and e-mail is 
FREE!!
>REPORT #2 and #5 will show you the best methods for bulk
>e-mailing, tell you where to obtain free bulk e-mail software and
>where to obtain e-mail lists and show you how to send out
>1,000,000 e-mails for free.
> >>>>>>>>>
> >>>>>>>>>METHOD #2 - PLACING FREE ADS ON THE INTERNET
> >>>>>>>>>
> >>>>>>>>>1.  Advertising on the 'Net is very, very  inexpensive,
>and there are HUNDREDS of FREE places to  advertise. Let's say 
you
>decide to start small just to see how well it
>works. Assume your goal is to get ONLY 6 people to participate
>on your first level. (Placing a lot of FREE ads on the internet
>will EASILY get a larger response.) Also assume that everyone 
else
>in YOUR ORGANIZATION gets ONLY 6 downline members.
>Follow this example to achieve the STAGGERING results below.
> >>>>>>>>>
> >>>>>>>>>1st level--your 6 members with   $5 ($5 x
>6).............$30
> >>>>>>>>>2nd level--6 members from those 6 ($5 x
>36).............$180
> >>>>>>>>>
> >>>>>>>>>3rd level--6 members from those 36 ($5 x 216).......
>$1,080
> >>>>>>>>>4th level--6 members from those 216 ($5 x 1,296)..
>$6,480
> >>>>>>>>>5th level-6 members from those 1,296 ($5 x 7,776)...
>$38,880
>
>-----------
> >>>>>>>>>             THIS TOTALS
>----------------------$46,650
> >>>>>>>>>
>Remember friends, this assumes that the people who participate
>only recruit 6 people each.  Think for a moment what would
>happen if they got 20 people to participate!  Many people will 
get
>100's of participants! THINK ABOUT IT!
>For every $5.00 you receive, all you must do is e-mail them
>the report they ordered.  THAT'S IT!  ALWAYS PROVIDE SAME-DAY
>SERVICE ON ALL ORDERS!  This will guarantee that
>the e-mail THEY send out, with YOUR name and address on it, will
>be prompt because they can't advertise until they
>receive the report!
> >>>>>>>>>
> >>>>>>>>>------------------------------------------
> >>>>>>>>>AVAILABLE REPORTS
> >>>>>>>>>------------------------------------------
> >>>>>>>>>
> >>>>>>>>>*** Order Each REPORT by NUMBER and NAME ***
> >>>>>>>>>
> >>>>>>>>>Notes:
> >>>>>>>>>
>-  ALWAYS SEND $5 CASH (U.S. CURRENCY) FOR EACH REPORT
>    CHEQUES NOT ACCEPTED
>-  ALWAYS SEND YOUR ORDER VIA FIRST CLASS MAIL-  Make sure the
>cash is concealed by wrapping it in at least two sheets of paper 
-
>  On one of those sheets of paper, include: (a) the number & name
>of the report you are ordering, (b) your e-mail address, and (c)
>your name & postal address.
> >>>>>>>>>
> >>>>>>>>>PLACE YOUR ORDER FOR THESE REPORTS NOW:
> >>>>>>>>>______________________________________________________
> >>>>>>>>>
> >>>>>>>>>
>REPORT #1  "The Insider's Guide to Advertising for Free on the
>Internet"
> >>>>>>>>>
>          ORDER REPORT #1 FROM:
> >>>>>>>>>
> >>>>>>>>>EAS Enterprises
> >>>>>>>>>196 Lehigh Circle
> >>>>>>>>>Tobyhanna, PA. 18466
> >>>>>>>>>______________________________________________________
> >>>>>>>>>
>REPORT #2  "The Insider's Guide to Sending Bulk E-mail on the
> >>>>>>>>>Internet"
> >>>>>>>>>
> >>>>>>>>>ORDER REPORT #2 FROM:
> >>>>>>>>>
> >>>>>>>>>American United Systems
> >>>>>>>>>3915 Mission Avenue Suite 7414
> >>>>>>>>>Oceanside, CA. 92054-7801
> >>>>>>>>>______________________________________________________
> >>>>>>>>>
>REPORT #3  "The Secrets to Multilevel Marketing on the Internet"
> >>>>>>>>>
> >>>>>>>>>ORDER REPORT #3 FROM:
> >>>>>>>>>
> >>>>>>>>>MLH Enterprises
> >>>>>>>>>1450 N. Santa Fe  Suite C-525
> >>>>>>>>>Vista, Ca. 92083
> >>>>>>>>>______________________________________________________
> >>>>>>>>>
>REPORT #4 "How to become a Millionaire utilizing the Power
> >>>>>>>>>of Multilevel Marketing and the Internet"
> >>>>>>>>>
> >>>>>>>>>ORDER REPORT #4 FROM:
> >>>>>>>>>
> >>>>>>>>>ROJ Inc.
> >>>>>>>>>PO BOX 5052
> >>>>>>>>>Oceanside, Ca. 92051-5052
> >>>>>>>>>
> >>>>>>>>>______________________________________________________
> >>>>>>>>>
>REPORT #5 "How to SEND 1,000,000 e-mails for FREE"
> >>>>>>>>>
> >>>>>>>>>ORDER REPORT #5 FROM:
> >>>>>>>>>
> >>>>>>>>>Marketing Unlimited
> >>>>>>>>>PO Box 2312
> >>>>>>>>>Carlsbad, Ca. 92008
> >>>>>>>>>______________________________________________________
> >>>>>>>>>
> >>>>>>>>>______________________________________________________
> >>>>>>>>>
>There currently more than 175,000,000 people online worldwide!
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>******* TIPS FOR SUCCESS *******
> >>>>>>>>>
> >>>>>>>>>*  TREAT THIS AS YOUR BUSINESS!  Be prompt, 
professional,
> >>>>>>>>>    and follow the directions accurately.
> >>>>>>>>>
> >>>>>>>>>*  Send for the five reports IMMEDIATELY so you will 
have
>them when the orders start coming in because: When you receive a
>$5 order, you MUST send out the requested product/report.
> >>>>>>>>>
> >>>>>>>>>*  ALWAYS PROVIDE SAME-DAY SERVICE ON THE ORDERS YOU
>RECEIVE.
>
> >>>>>>>>>*  Be patient and persistent with this program. If you
>follow the instructions exactly, your results WILL BE SUCCESSFUL!
> >>>>>>>>>
>*  ABOVE ALL, HAVE FAITH IN YOURSELF AND KNOW YOU WILL SUCCEED!
>
> >>>>>>>>>
> >>>>>>>>>******* YOUR SUCCESS GUIDELINES *******
> >>>>>>>>>
> >>>>>>>>>Follow these guidelines to guarantee your success:
> >>>>>>>>>If you don't receive 20 orders for REPORT #1 within two
>weeks, continue advertising or sending e-mails until you do.
>Then, a couple of weeks later you should receive at least 100
>orders for REPORT#2.  If you don't, continue advertising or
>sending e-mails until you do. Once you have received 100 or more
>orders for REPORT #2, YOU CAN RELAX, because the system is 
already
>working for you, and the cash will continue to roll in!
> >>>>>>>>>
> >>>>>>>>>THIS IS IMPORTANT TO REMEMBER:
> >>>>>>>>>
> >>>>>>>>>Every time your name is moved down on the list, you are
>placed in front of a DIFFERENT report.  You can KEEP TRACK of 
your
>PROGRESS by watching which report people are ordering from you. 
If
>you want to generate more income, send another batch of e-mails 
or
>continue placing ads and start the whole process again!
> >>>>>>>>>There is no limit to the income you will generate from
>this business!
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>Before you make your decision as to whether or not you
>participate in this program.  Please
>answer one question.  DO YOU WANT TO CHANGE YOUR LIFE?   If the
>answer is yes, please look at the following facts about this
>program:
> >>>>>>>>>
> >>>>>>>>>   1. YOU ARE SELLING A PRODUCT WHICH DOES NOT
> >>>>>>>>>       COST ANYTHING TO PRODUCE!
> >>>>>>>>>
> >>>>>>>>>   2. YOU ARE SELLING A PRODUCT WHICH DOES NOT
> >>>>>>>>>       COST ANYTHING TO SHIP!
> >>>>>>>>>
> >>>>>>>>>   3. YOU ARE SELLING A PRODUCT WHICH DOES NOT
> >>>>>>>>>       COST YOU ANYTHING TO ADVERTISE!
> >>>>>>>>>
> >>>>>>>>>   4. YOU ARE UTILIZING THE POWER OF THE INTERNET
> >>>>>>>>>       AND THE POWER OF MULTI-LEVEL MARKETING TO
> >>>>>>>>>       DISTRIBUTE YOUR PRODUCT ALL OVER THE
> >>>>>>>>>       WORLD!
> >>>>>>>>>
> >>>>>>>>>   5.  YOUR ONLY EXPENSES OTHER THAN YOUR
> >>>>>>>>>       INITIAL $25  INVESTMENT IS YOUR TIME!
> >>>>>>>>>
> >>>>>>>>>   6.  VIRTUALLY ALL OF THE INCOME YOU GENERATE
> >>>>>>>>>        FROM THIS PROGRAM IS PURE PROFIT!
> >>>>>>>>>
> >>>>>>>>>   7.  THIS PROGRAM  WILL CHANGE YOUR
> >>>>>>>>>        LIFE FOREVER.
> >>>>>>>>>
> >>>>>>>>>******* T  E  S  T  I  M  O  N  I  A  L  S *******
> >>>>>>>>>
> >>>>>>>>>This program does work, but you must follow it EXACTLY!
>Especially the rule of not trying to place your name in a
>different position, it won't work and you'll lose a lot of
>potential income.  I'm living proof that it works. It really is
>a great opportunity to make relatively easy money, with little
>cost to you.  If you do choose to participate, follow the
>program exactly, and you'll be on your way to financial
>security.
> >>>>>>>>>                     Fred Dellaca, Westport, New 
Zealand
> >>>>>>>>>
>       My name is Mitchell. My wife, Jody, and I live in Chicago,
>IL. I am a cost accountant with a major U.S. Corporation and I
>make pretty good money. When I received the program I grumbled
>to Jody about receiving "junk mail." I made fun of the whole
>thing, spouting my knowledge of the population and percentages
>involved.  I "knew" it wouldn't work. Jody totally ignored my
>supposed intelligence and jumped in with both feet. I made
>merciless fun of her, and was ready to lay the old "I told you
>so" on her when the thing didn't work... well, the laugh was on
>me! Within two weeks she had received over 50 responses. Within
>45 days she had received over $147,200 in $5 bills! I was
>shocked!  I was sure that I had it all figured and that it
>wouldn't work.  I AM a believer now. I have joined Jody in her
>"hobby."   I did have seven more years until retirement, but I
>think of the "rat race" and it's not for me. We owe it all to
>MLM.
> >>>>>>>>>                     Mitchell Wolf MD., Chicago, IL
> >>>>>>>>>
>      The main reason for this letter is to convince you that 
this
>system is honest, lawful, extremely profitable, and is a way to
>get a large amount of money in a short time. I was approached
>several times before I checked this out. I joined just to see
>what one could expect in return for the minimal effort and money
>required.  To my astonishment, I received $36,470.00 in the
>first 14 weeks, with money still coming in.
> >>>>>>>>>
> >>>>>>>>>                            Sincerely yours,
> >>>>>>>>>           Pam Hedland   Halmstad, Sweden
> >>>>>>>>>
>      Not being the gambling type, it took me several weeks to
>make up my mind to participate in this plan. But conservative
>that I am, I decided that the initial investment was so little
>that there was just no way that I wouldn't get enough orders to
>at least get my money back. I surprised when I found my
>medium-size post office box crammed with orders! For awhile, it
>got so overloaded that I had to start picking up my mail at the
>window. I'll make more money this year than any 10 years of my
>life before. The nice thing about this deal is that it doesn't
>matter where people live. There simply isn't a better investment
>with a faster return.
> >>>>>>>>>                  Dan Sondstrom, Alberta, Canada
> >>>>>>>>>
> >>>>>>>>>     I had received this program before. I deleted it,
>but later I wondered if I shouldn't have given it a try. Of
>course, I had no idea who to contact to get another copy, so I 
had
>to wait until I was e-mailed another program, ...11 months passed
>then it came...I didn't delete this one!...I made more than
>$41,000 on the first try!!
> >>>>>>>>>                     Mohamed, Cairo, Egypt
> >>>>>>>>>
> >>>>>>>>>      This is my third time to participate in this 
plan.
>We have quit our jobs, and will soon buy a home on the beach and
>live off the interest on our money. The only way on earth that
>this plan will work for you is if you do it. For your sake, and
>for your family's sake don't pass up this golden opportunity. 
Good
>luck and happy spending!
> >>>>>>>>>                    Sam Lee  Suva, Fiji Islands
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>ORDER YOUR REPORTS TODAY AND
> >>>>>>>>>GET STARTED ON YOUR ROAD TO
> >>>>>>>>>FINANCIAL FREEDOM!
> >>>>>>>>>
> >>>>>>>>>NOW  IS THE TIME FOR YOUR TURN
> >>>>>>>>>
> >>>>>>>>>DECISIVE ACTION YIELDS
> >>>>>>>>>POWERFUL RESULTS
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>PLEASE NOTE:  If you need help with starting a 
business,
>registering a business name, learning how income tax is handled,
>etc., contact your local office of the Small Business
>Administration (a Federal agency) 1-(800)827-5722 for free help
>and answers to questions.  Also, the Internal Revenue Service
>offers free help via telephone and free seminars about business
>tax requirements.  Your earnings and results are highly dependant
>on your activities and advertising.  This letter constitutes no
>guarantees stated nor implied.  In the event that it is 
determined
>that this letter constitutes a guarantee of any kind, that
>guarantee is now void.   Any testimonials or amounts of earnings
>listed in this letter may be factual or fictitious.  If you have
>any question of the legality of this letter contact the Office of
>Associate Director for Marketing Practices Federal Trade
>Commission Bureau of Consumer Protection in WashingtonDC.
 
 
 
 
 
 
 
 
 
 
 
 



From rem-conf-request@es.net  Tue Oct  5 20:29:17 1999
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Date: Tue, 05 Oct 1999 17:16:08 -0700
From: Don Brutzman <brutzman@nps.navy.mil>
Organization: Naval Postgraduate School
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To: AVT working group <rem-conf@es.net>,
        JMF Java media framework list <JMF-INTEREST@java.sun.com>,
        vrtp working group <vrtp@web3d.org>, movesrg <moves@cs.nps.navy.mil>,
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Subject: rtpMonitor thesis and software available
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         IMPLEMENTING A MONITOR APPLICATION FOR THE
        REAL-TIME TRANSPORT PROTOCOL (RTP) USING THE
                 JAVA MEDIA FRAMEWORK (JMF)

                  Francisco Carlos Afonso
           Lieutenant Commander, Brazilian Navy
    Naval Postgraduate School, Monterey California USA
                    September 1999

Abstract.  The Real-time Transport Protocol (RTP) supports the 
transmission of time-based media, such as audio and video, 
over wide-area networks (WANs), by adding synchronization 
and quality-of- service (QoS) feedback capabilities to the 
existing transport protocol. RTP has been widely used in 
the Multicast Backbone (MBone), a virtual network that has 
become a shared worldwide medium for Internet multicast
communications.

	This work presents the design patterns, architecture 
and implementation of an RTP monitor application using the 
Java Media Framework (JMF), a new Java Application Programming 
Interface (API) for multimedia support. An RTP monitor is an
application that receives packets from all participants in a 
multicast session in order to estimate the quality of service 
for distribution monitoring, fault diagnosis and both short
and long-term statistics.

	This new RTP monitor is available as a component of 
the Virtual Reality Transfer Protocol (vrtp), a protocol 
being developed to support large-scale virtual environments
(LSVEs) over the Internet. Initial test results are 
satisfactory for audio and video streams, as well as prototype 
RTP-compliant Distributed Interactive Simulation (DIS) protocol
streams. Future work includes automated monitoring across WANs 
and standardizing structured data formats to comply with 
Management Information Base (MIB) requirements using Extensible 
Markup Language (XML) target set definitions.


The thesis and software are available at
http://www.web3D.org/WorkingGroups/vrtp/rtpMonitor/rtpMonitor.html
Sample screen snapshots are available at
http://www.web3D.org/WorkingGroups/vrtp/rtpMonitor/manual.html

Comments, bricks and bouquets are welcome.

all the best, Don
-- 
Don Brutzman  Naval Postgraduate School, Code UW/Br Root 200  work 831.656.2149
              Monterey California 93943-5000 USA              fax  831.656.3679
Virtual worlds/underwater robots/Internet     http://web.nps.navy.mil/~brutzman



From rem-conf-request@es.net  Wed Oct  6 02:24:36 1999
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Reply-To: <gunnar.hellstrom@omnitor.se>
From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: <ietf-drafts@ietf.org>
Cc: "IETF-avt list" <rem-conf@es.net>
Subject: New version of text conversation RTP draft
Date: Wed, 6 Oct 1999 07:41:55 +0200
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This is a submission of a new version of the RTP-Text internet draft:
draft-ietf-avt-rtp-text-01.txt

This version contains minor editorial adjustments and added clarification=
s
in the recommended procedures part, and an updated address of the author.

Regards
______________________________________
Gunnar Hellstr=F6m
Ericsson Mobile Communication AB
Sweden

Tel +46 708 204 288
Fax +46 8 556 002 06
Video +46 8 556 002 05
Txt (All kinds) +46 8 556 002 05
E-mail gunnar.hellstrom@omnitor.se

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Internet Engineering Task Force                                   AVT WG
Internet Draft                                          Gunnar =
Hellstr=F6m
draft-ietf-avt-rtp-text-01.txt                                  Ericsson
October  6, 1999
Expires: April 6, 2000


                      RTP Payload for Text Conversation

STATUS OF THIS MEMO

   This document is an Internet-Draft and it is in full conformance with
   all provisions of section 10 of RFC2026. Internet-Drafts are working
   documents of the Internet Engineering Task Force (IETF), its areas,
   and its working groups.  Note that other groups may also distribute
   working documents as Internet-Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as ``work in progress''.

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/lid-abstracts.txt

   The list of Internet-Draft Shadow Directories can be accessed at=20
   http://www.ietf.org/shadow.html

   Distribution of this document is unlimited.


ABSTRACT

   This memo describes how to carry text conversation session contents
   in RTP packets. Text conversation session contents is specified in
   ITU-T Recommendation T.140 [1].=20

   Text conversation is used alone or in connection to other
   conversational facilities such as video and voice, to form multimedia
   conversation services.

   This RTP payload description contains an optional possibility to
   include redundant text from already transmitted packets in order to
   reduce the risk of text loss caused by packet loss. The redundancy
   coding follows RFC 2198.






Hellstrom                                                       [Page 1]
=0C
Internet Draft


1 Introduction

   This memo defines a payload type for carrying text conversation
   session contents in RTP packets. Text conversation session contents
   are specified in ITU-T Recommendation T.140 [1]. Text conversation is
   used alone or in connection to other conversational facilities such
   as video and voice, to form multimedia conversation services. Text in
   text conversation sessions is sent as soon as it is available, or =
with
   a small delay of up to 0.5 seconds for buffering.
   The text is supposed to be entered by human users from a keyboard,
   handwriting recognition, voice recognition or any other input method.
   The rate of character entry is usually at a level of a few characters
   per second or less. Therefore, the expected number of characters to
   transmit is low. Only one or a few new characters are expected to be
   transmitted with each packet.

   T.140 specifies that text and other T.140 elements SHALL be
   transmitted in ISO 10 646-1 code with UTF-8 transformation. That
   makes it easy to implement internationally useful applications, and
   to handle the text in modern information technology environments.
   The most common T140 PDU is a character of ISO 10646 text, UTF-8
   coded, submitted from T.140 without any extra framing.=20
=20
   T.140 requires the transport channel to provide characters without
   duplication and in original order.=20
   Text conversation users expect that text will be delivered with no or
   a low level of lost information. If lost information can be replaced
   with a special marker, the willingness to accept loss is expected to
   be higher.
=20
   Therefore a mechanism based on RTP is specified here. It gives text
   arrival in correct order, without duplications, with an optional
   possibility to repeat data for redundancy to lower the risk of loss,
   and a mechanism that reveals loss and therefore can insert a marker
   for lost text in the received stream. Since packet overhead is
   usually much larger than the T.140 contents, the increase in channel
   load by the redundancy scheme is minimal.=20

1.1 Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [4]

2. Usage of RTP

   When an unreliable transport of T.140 text session data in a packet
   network is selected, RTP with payload as described in this
   specification should be used. T.140 data is submitted for =
transmission
   in a number of complete protocol data units (T140 PDU). A T140block
   contains one or more T140 PDU:s.=20
  =20

Hellstr=F6m                                                       [Page =
2]
=0C
Internet Draft                                 =20


   A text conversation RTP packet contains an RTP header, a Payload
   Header, OPTIONAL redundant data fields, and the new (primary)
   T.140 data field.

2.1 RTP packet header
  =20
   Each RTP packet starts with a fixed RTP header. The following fields
   of the RTP fixed header are used for T.140 text streams:

Payload Type (PT): Payload type allocation is dynamic. If redundancy is
   used, the Payload Type SHALL indicate redundancy ("RED") according
   to RFC 2198 [3]. If no redundancy is used, the Payload Type SHALL
   specify the primary T.140 data "T140" payload format.=20

Sequence number:  The Sequence Number SHALL be increased with one for
   each new transmitted packet. It is used for detection of packet loss
   and packets out of order, and can be used in the process of retrieval
   of redundant text, reordering of text and marking missing text.

Timestamp: The RTP Timestamp encodes the approximate sampling instance
   of the primary text in the packet. A clock frequency of 1000 Hz SHALL
   be used. No sequential packets SHOULD use the same timestamp.=20

2.2 Additional headers

   When redundant data is used, additional headers specify the redundant =

   text fields. They are specified according to RFC 2198 and use dynamic =

   payload type "T140".

2.3 T.140 Text structure

   T.140 text is UTF-8 coded as specified in T.140 with no extra =
framing.
   When using the format with redundant data, the transmitter MAY
   select a number of T.140 block generations to retransmit in each
   packet. A higher number introduces better protection against loss of
   text. It is RECOMMENDED to use no more than 6 generations. The=20
   redundant data is placed in age order with most recent redundant text
   last in the redundance area.
 =20

3. Recommended procedures

   This section contains RECOMMENDED procedures for usage of the payload
   format.
   Based on the information in the received packets, the receiver can:
      - reorder received text out of order.
      - mark where text is missing because of packet loss.
      - compensate for lost packets by using redundant data.




Hellstr=F6m                                                       [Page =
3]
=0C
Internet Draft                  		=09


3.1 Recommended basic procedure  =20

   On reception, the RTP sequence number is compared with the sequence
   number of the last correctly received packet. If they are =
consecutive,
   only the most recent t140block is retrieved from the received packet. =



3.2 Recommended procedure for compensation for lost packets.

   For reduction of data loss in case of packet loss, redundant data MAY
   be included in the packets.
   If network conditions are not known, it is RECOMMENDED to use two
   generations of T.140 blocks in the packets. If there is a gap in
   the RTP sequence numbers, and redundancy coded T.140blocks are
   available in the packet, older t140blocks are retrieved from the
   packet up to the point where the sequence number is consecutive to
   the last correctly received packet or no more t140block are available
   in the received packet.=20

   Both for the case when redundancy is used and not used, missing data
   SHALL be marked. If there is still a gap in the sequence, one T.140
   missing data marker ( Unicode character 2607 "Lightning") SHALL be
   inserted, UTF-8 coded, replacing each missing t140block.=20

3.3 Recommended procedure for compensation for packets out of order.

   For protection against packets arriving out of order, the following
   procedure MAY be implemented in the receiver.
   If analysis of a received packet reveals gaps in the sequence,
   the received packet can be kept in a buffer before being used as
   valid T.140 data. If a packet arrives with a t140block belonging to
   the gap, the gap is filled with the contents of this block. If all
   gaps are filled or the packet is kept in the buffer for 0.5 seconds,
   valid T140blocks in the buffered packet are retrieved and used as
   valid T.140 data. For each t140block still missing, a T.140 missing
   data marker (Unicode character 2607 "Lightning") is inserted.=20

3.4 Transmission during "silent periods" when redundancy is used.

   When using the redundancy transmission scheme, and there is nothing
   more to transmit from T.140, the latest t140block has a risk of
   getting old before it is transmitted as redundant data. The result
   is less useful protection against packet loss at the end of a text
   input sequence. For cases where this should be avoided,
   the ISO 10 646 synchronization character FEFF MAY be transmitted
   (UTF-8 coded) after some time of inactivity.=20









Hellstr=F6m                                                       [Page =
4]
=0C
Internet Draft



4. Examples

   This is an example of a T140 RTP packet without redundancy.
    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=3D2|P|X| CC=3D0  |M| T140 dyn PT |   sequence number             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              timestamp (1000Hz)                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   +                T140 encoded data (PT=3Ddynamic)                 +
   |                                                               |
   +                                               +---------------+
   |                                               |              =20
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+              =20

    This is an example of an RTP packet with two t140 blocks using
    redundancy coding.
    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=3D2|P|X| CC=3D0  |M| "RED" PT    |   sequence number of primary  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              timestamp  of primary encoding "P"               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |1| T140 dyn. PT|  timestamp offset of "R"  | "R" block length  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |0| T140 dyn. PT|                                               |
   +-+-+-+-+-+-+-+-+                                               +
   |                                                               |
   +            "R" T140 encoded redundant data (PT=3Ddynamic T140)  +
   |                                                               |
   +                                               +---------------+
   |                                               |               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               +
   |             "P" T140 encoded primary data (PT=3Ddynamic T140)   |
   +                                                               +
   +                                               +---------------+
   |                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Figure: Examples of RTP text packets.






Hellstr=F6m                                                       [Page =
5]
=0C
Internet Draft

5.  Security Considerations.

    Since the intention of the described payload format is to carry text
    in a text conversation, security measures in the form of encryption
    is of importance. The amount of data in a text conversation session
    is low and therefore any encryption method MAY be selected and
    applied to T.140 session contents or to the whole RTP packets.=20
    When redundant data is included, the same security considerations
    as for RFC 2198 apply.

6  MIME Media Type Registrations

    This document defines a new RTP payload name and associated MIME
    type, T140 (text/t140). The registration form for this MIME type has
    been submitted to IANA.
  =20
6.1  Registration of MIME media type text/t140

     MIME media type name: text

     MIME subtype name: t140

     Required parameters: None

     Optional parameters: None

     Encoding considerations: T140 text can be transmitted=20
     with RTP as specified in "draft-ietf-avt-rtp-text-01".

     Security considerations: None

     Interoperability considerations: None

     Published specification: ITU-T T.140 Recommendation.
                              draft-ietf-avt-rtp-text-01.    =20

     Applications which use this media type:
       Text communication terminals and text conferencing tools.

     Additional information: None

       Magic number(s): None
       File extension(s): None
       Macintosh File Type Code(s): None

     Person & email address to contact for further information:
       Gunnar Hellstrom
       e-mail: gunnar.hellstrom@omnitor.se

     Intended usage: COMMON

     Author/Change controller:
       Gunnar Hellstrom
       e-mail: gunnar.hellstrom@omnitor.se


Hellstr=F6m                                                       [Page =
6]
=0C
Internet Draft                                Expires: April 6, 2000


7.  Author's Address

   Gunnar Hellstrom
   Ericsson Mobile Communication AB
   Home Communications
   SE-164 80 Stockholm  =20
   Sweden

   e-mail: gunnar.hellstrom@omnitor.se
   Tel:    +46 708 204 288
   Fax:    +46 8 556 002 06

8 References

[1] ITU-T Recommendation T.140 (1998) - Text conversation protocol for
    multimedia application, with amendment 1999.

[2] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson,=20
   "Real Time Transport Protocol", RFC 1889.

[3] C. Perkins, I. Kouvelas, V. Hardman, M. Handley, and J. Bolot,
   "RTP payload for redundant audio data," RFC 2198, Internet
   Engineering Task Force, Sept. 1997.

[4] S. Bradner, "Key words for use in RFCs to Indicate Requirement
    Levels", RFC 2119, March 1997.

9   Revision history.

    Oct 6 1999. draft-01. Explanations in recommended procedures =
improved.























Hellstr=F6m                                                       [Page =
7]
=0C

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From rem-conf-request@es.net  Wed Oct  6 03:16:26 1999
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Date: Wed, 6 Oct 1999 12:25:50 +0530 (IST)
From: Rafiq Shaikh <rafiq@chiplogic.com>
To: rem-conf@es.net
Subject: Test tools for RTP/RTCP
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Hi all,

Are there any s/w test tools available to test the functionality of the 
RTP/RTCP s/w?

any pointers will be very helpful.

Thanks in advance.

-Rafiq.





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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload Format for Real-Time Pointers
	Author(s)	: M. Civanlar, G. Cash
 	Filename	: draft-ietf-avt-pointer-00.txt,.ps
	Pages		: 5
	Date		: 05-Oct-99
	
This document describes an RTP [1] payload format for transporting the
coordinates of a dynamic pointer that may be used during a
presentation. Although a mouse can be used as the pointer, this
payload format is not intended and may not have all functionalitiesp
needed to implement a  general mouse event transmission mechanism.

A URL for this Internet-Draft is:
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Apologize for the wide cross-posting
----------------------------------------------------------------------------
-------------------
CALL FOR PAPERS
Engineers Conference on Broadband Internet Access Technologies, Systems and
Services
May 10-11, 2000 - Las Vegas Conventions Center as  part of  the
Networld+Interop'2000 Program
For the detailed call for papers and the program see:
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Paper Submission Deadline - December 1, 1999
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cross-posting</FONT>
<BR><FONT SIZE=3D2 =
FACE=3D"Arial">---------------------------------------------------------=
--------------------------------------</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">CALL FOR PAPERS</FONT>
<BR><FONT FACE=3D"Times New Roman">Engineers Conference on Broadband =
Internet Access Technologies, Systems and Services</FONT>
<BR><FONT FACE=3D"Times New Roman">May 10-11, 2000 - Las Vegas =
Conventions Center as&nbsp; part of&nbsp; the Networld+Interop'2000 =
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<P><FONT SIZE=3D2 FACE=3D"Arial">Paper Submission Deadline - December =
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From rem-conf-request@es.net  Fri Oct  8 05:15:21 1999
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From: "Lars-Erik Jonsson" <Lars-Erik.Jonsson@ericsson.com>
To: "Ali Irfan-FIA225" <fia225@email1.wes.mot.com>, <pilc@grc.nasa.gov>,
        <rem-conf@es.net>
Cc: "Natarajan Nat-ANN004" <ann004@email.mot.com>
Subject: Re: Header Stripping
Date: Fri, 8 Oct 1999 10:45:45 +0200
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Hello !

[To: PILC and AVT, AVT included since original message appered at the
PILC-list]

Ali, the answers to your questions should be ABSOLUTELY NOT and PROBABLY
NOT.

Header stripping is in general something really ugly and such ideas should
not be accepted. However, there is a reason for why ideas like that come up.
The reason is that in cellular environments, the header compression scheme
for real-time (e.g. CRTP) of today are not working very well due to the
problematic characteristics of the links. This was studied in the internet
draft "draft-degermark-crtp-cellular". New header compression solutions are
needed that are efficient and robust enough against packet losses and
thereby making things like header stripping uninteresting.

A new concept for header compression, ROCCO, has been proposed and the work
can be studied at:
www.ludd.luth.se/users/larsman/rocco

It is important to speed up the progress of the header compression
development and standardization before ideas like those about header
stripping has come to far. Support for and participation in the ROCCO
development is needed

/Lars-Erik Jonsson


>Question:
>1. Should a recommendation for header stripping/regeneration be allowed
>in any IP network?
>2. Also, should this issue be dealt with in the performance
>implications draft?
>
>Background:
>
>I was just reading a Cellular network proposal (the name of group or doc.
is
>not important) for an IP network. For air-link optimization,
>header-compression is recommend. Fine. In addition an option for
>header-striping and regeneration down-stream is also allowed.  A statement
>in the doc. states,
>
> "Reducing of the header size is done by removing redundancy in the
>originally coded header information AND/OR REMOVING HEADER FIELD
INFORMATION
>AND THERBY LOSING FUNCTIONALITY"
>
>This disturbed me. I want to make a contribution to the group to delete
such
>text, but would first like  to get some feedback from this mailing list.
>
>Note: the doc. does state that these options are to be investigated and a
>final recommendation made. So header-stripping option is not set in stone,
>yet.
>
>Irfan Ali
>Motorola




From rem-conf-request@es.net  Fri Oct  8 15:33:25 1999
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Date: Fri, 8 Oct 1999 15:07:30 -0400
From: jerome@psti.com
To: rem-conf@es.net
Subject: RTP testing
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Hello,

i'm implementing RTP and just read draft-ietf-avt-rtptest-01.txt
about "RTP testing strategie". I would like to know if there is a 
available software able to perform these tests automatically.

thanks



From rem-conf-request@es.net  Sat Oct  9 06:38:48 1999
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DATE: 09 Oct 99 3:08:20 AM
FROM: jobaldi@ibm.net
Reply-to: jobaldi@ibm.net
TO: jobaldi@ibm.net
SUBJECT: 57 Million Email Addresses - ONLY $149.00
Message-Id: <E11ZtL5-00035E-00@mail1.es.net>
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If you can not see the just how excellent the

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To make money you must stop dreaming 

and TAKE ACTION.


****************************************

THE BRONZE MARKETING SETUP

57,000,000 email addresses on CD

These name are all in text files

ready to mail!!!

$149.00

****************************************

THE SILVER MARKETING SETUP

57,000,000 email addresses on CD

These name are all in text files

ready to mail!!!       AND

Several different email programs 

and tools  to help with your mailings

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****************************************

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These name are all in text files

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Several different email programs 

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             AND 

Over 500 different Business Reports

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With this package you get the email addresses, 

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AND ...... 

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$ 249

*************************************************************
SEVERAL WAYS TO ORDER !!!

IF YOU ORDER BY PHONE OR FAX WE WILL SHIP YOUR CD CONTAINING THE 57 MILLION + NAMES WITHIN 12 HOURS OF YOUR ORDER!!!

 
  1) WE ACCEPT:  AMERICAN EXPRESS OR
    
                               VISA <> MasterCard
 
     TYPE OF CARD  AMX / VISA / MC??_______________

      EXPIRATION DATE   ___________________________
 
     NAME ON CREDIT CARD________________________
 
     CREDIT CARD #________________________________
 
     BILLING ADDRESS ____________________________
 
     CITY_________________________________________
 
    STATE________________ZIP_____________________
 
   PHONE INCLUDE AREA CODE___________________
   
    EMAIL ADDRESS______________________________ 

        WE WILL BILL selected amount to your account plus the following shipping costs

        SHIPPING  COST OF 3.85 FIRST CLASS MAIL

        SHIPPING COST OF  15.00  24 HOUR EXPRESS MAIL /  FEDERAL EXPRESS
 
        SALES TAX  added where required 
 
FAX your order and a copy of your SIGNED check payable to Cyberdata 

 for the appropriate amount to;

530-895-8470


   2) Fax the same above requested credit card 

information to 530-895-8470. 
   

   3) Call phone # 530-872-6591.  This is a 24 hour phone 

number to place a CREDIT CARD order. This is an ORDER LINE only.

ALL INFORMATION NECESSARY FOR YOU TO SUCCESSFULLY MAIL, QUICKLY, PROPERLY, LEGALLY PROVIDED WITH ORDER


Copyright 1999

All rights reserved









From rem-conf-request@es.net  Sun Oct 10 13:16:16 1999
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Subject: $E-commerce will increase 40 times by 2003!
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EARN  LIMITLESS RESIDUAL INCOME FROM INTERNET COMMERCE!!

CALL 888-277-7691 TODAY!

We are seeking qualified representatives for our new Internet-based
technology company, which is financed by a well-known, 15-year-old
public corporation.

This parent corporation generates about $1.5 billion-per-year in
gross revenue, and its top representatives earn from $1,000,000
to almost $3,000,000 per year!

Conservative estimates predict that in 5 - 10 years our technology
company will generate between $10 and $20 billion per year, with
our TOP REPRESENTATIVES earning ALMOST UNIMAGINABLY HIGH
INCOMES of at least TEN TIMES GREATER than those of the
parent company!!

Many INTERNET FORTUNES will be created over the next 5 years..

ONE OF THEM MIGHT AS WELL BE YOURS!!!

Almost everyone who is aware of the rapid growth of the Internet
has at some time envisioned themselves:

- Making huge profits from an Internet-based business,
- Working entirely from home,
- Spending quality time with their families, and
- Forever leaving behind the boss, the office, and the daily commute.
- If you have too, then please read on!

CONSIDER THE FOLLOWING:

- Today, only 18% of North Americans have Internet access in their homes.

- However, Internet use is doubling every 100 days!  In it's first 
major study of the economic effect of the Internet, the U.S. Commerce
Department said that Internet traffic is doubling every 100 days and
Electronic Commerce should reach $300 Billion by 2002!

- There are 85,000 new Internet customers logging on daily!

- Many different sources have predicted that Internet Commerce will 
increase at least twenty-fold in the next 5 years, and that within 10 
years, standard retailing of products will largely be replaced by 
online selling!

- Experts are predicting that computer sales will drop drastically
over the next few years because most homes will be logging onto the
net using simpler, less expensive Internet Access Devices.

- Some companies are already generating massive revenue by directing 
their customers to welcome or portal pages that display ads and banners
tempting them to shop online.  For example, during 1998, America Online
generated 3.2 billion dollars of revenue through its entry page.

ADDITIONALLY:

Market research has shown the more services (such as Internet, long
distance, cellular phone, and paging services) one company can
provide to the same customer, the longer they are likely to
retain that customer.

- If they provide one service, they have less than a 50% chance
of keeping that customer for one year.
- However, if they provide two services, they have a 75% chance
of keeping that customer for 5 years.
- Finally, of a company can provide three or more services to one
customer, they have a 90% chance of keeping that customer for
AT LEAST 10 YEARS!

THEREFORE:

The race is on! Today all major telecommunications companies are competing to:

- Secure their share of the Internet marketplace by acquiring
new Internet users and capturing their online dollars,
- Keep those customers for the next several years by offering
them multiple services in addition to Internet access, and
- Ultimately generate huge profits when E-Commerce explodes and their
customer base spends 40 dollars for every 1 they are spending today!

NOW, IMAGINE IF YOU COULD:

- Sell Internet service.
- Sell your customers an Internet Access Device to connect to that service.
- Profit every month from every Internet account.
- Direct every customer, every time they log on, to your own Online
Supermall wehre they could purchase every imaginable product or service.
- Profit from every online purchase made by every customer.
- Earn enormous residual profits from all business generated by a
potentially huge network of others who you recruit to dupicate
your efforts, and thus
- Capture your share of the E-Commerce now and make a fortune as it grows.

"A wonderful idea for large company" you say, "but a fairy tale for me.."

NOT ANY MORE!

Our company is already "leveling the playing field" and can provide
all the resources and tools you need to help you start our Internet
Business and position yourself today to make a fortune from the
Internet explosion!

We are a rapidly-Growing technology company that is owned and financed
by a 15-year-old international billion-dollar company that is publicly
traded on the NYSE.  We offer a uniquely stable ground floor
opportunity, because our company is backed by an established business
with a Dunn & Bradstreet credit rating of 5A1, the highest rating
available to any business!!

We are currently seeking qualified individuals to independently market
our rapidly-growing nationwide Internet and telecommunications services
and our state-of-the-art Internet Access Device throughout the entire
United States and, in the near future, Canadian and Japanese markets!


WE ARE SEEKING INDIVIDUALS WHO ARE:

- Excited by the possibility of creating their own Internet fortune.
- Understand why they must begin establishing their Internet presence
today in order to do so.
- Are willing to take immediate action and use our extensive
resources and support to rapidly build their own business.
- Are able to invest between $500 and $1000 to start their business.
- Are serious about starting now!


PLEASE NOTE:

Our company is structured using a network marketing model similar to
that of our 15-year-old parent company.  This allows any individual
to start his or her own business without a huge initial investment
and makes our extensive resources and technology available to all who
wish to use them.  However, we only want serious, motivated individuals
to consider joining our company and do not want our network marketing
structure and relatively low start-up cost to attract the casually
interested.

HOWEVER:

If you ARE serious, this is TRULY THE CHANCE OF A LIFETIME (and perhaps
your only real chance) to "MAKE IT BIG" on the Internet!

We can provide you with the following products and technology to 
successfully compete in the Internet marketplace:

NATIONWIDE INTERNET ACCESS
We offer our customers state-of-the-art Internet access through a
high-speed backbone.  We have over 2500 dial-in numbers that make us
locally available anywhere in the country.

THE WORLD'S BEST-SELLING INTERNET SCREEN PHONE
We have exclusive rights to market the first and most popular Internet
Access Device in the world!  With this compact unit the average user
can be online, surfing the web, and sending and retrieving email within
10 minutes of taking it out of the box!!  This device is available
now and is selling rapidly.

AN EXTENSIVE E-COMMERCE WEBSITE
We are rapidly developing a comprehensive online store where our
customers will be able to purchase virtually any product or service
they need!  We already have hundreds of thousands of products and
services as well as direct links to several affiliate sites, and we
are adding new items every day!  All transactions are carried out
on our secure servers and all products are shipped promptly.

A PORTAL PAGE LEADING TO OUR E-COMMERCE WEBSITE
We have designed an entry page to the web so that every time any of
your customers or representatives logs on to the Internet to check
their email or to access a website, they view this page which displays
banners and ads directing them to our E-Commerce Store!  And, you
profit every time one of your customers or representatives makes an
online purchase!

MULTIPLE TELECOMMUNICATIONS SERVICES
We can keep our customers by offering them multiple, state-of-the-art
telecommunications services, including long distance telephone service,
cellular service, digital and alpha-numeric pager accounts, web
hosting, an online university, and more!  Furthermore, as more and more
services deregulate, such as local telephone service and utilities,
we will be competing for all of them!

A COMPREHENSIVE TRAINING WEBSITE
The two top representatives in our organization have designed a
detailed training site which clearly and objectively describes how to
build a successful business.  This website contains essential
information on prospecting, recruiting, training new members, and
acquiring new customers.  Most importantly, as your business grows,
it will dramatically decrease the time you spend training your
representatives and free valuable time for you to continue building
your organization.

SO, AS A REPRESENTATIVE, YOU CAN PROVIDE INDIVIDUALS WITH:

1) An Internet account.
2) A convenient device with which to access the web.
3) An E-Commerce site at which to shop.
4) Direction to that E-Commerce site every time they log on.
5) Multiple other telecommunications services.

ALL THE NECESSARY TOOLS YOU NEED TO:

- CAPTURE A SIGNIFICANT PERCENTAGE OF NEW INTERNET USERS,
- PROFIT FROM ALL E-COMMERCE THEY GENERATE, and
- RAPIDLY EXPAND YOUR BUSINESS TO KEEP UP WITH INTERNET GROWTH.

CLEARLY, IF YOU STARTED NOW AND OVER THE NEXT FEW YEARS
CAPTURED EVEN A TINY AMOUNT OF THE E-COMMERCE PROFITS
ON THE INTERNET, YOU WOULD CREATE A RESIDUAL INCOME OF
TRULY STAGGERING PROPORTIONS!!

REMEMBER!!
- Our parent company has, over the last 15 years, created annual incomes
of $152,098.00 to $649,550.15 for over 1,000 individuals.
- Its representatives have a millionaires' circle and a 10-million-
dollar circle.
- It did, as a whole, over $1.5 billion in sales last year.
- The revenue ultimately generated by our new company, which will
ride the ensuing tidal wave of Internet Commerce,
WILL LIKELY SURPASS THAT OF THE PARENT COMPANY BY AT LEAST 10 TIMES!!

DON'T MISS YOUR CHANCE TO PARTICIPATE IN THE FORTUNES
THAT ARE GOING TO BE CREATED!

LET US HELP YOU START BUILDING..

CALL TOLL FREE 1-888-277-7691 TODAY!!



From rem-conf-request@es.net  Sun Oct 10 23:00:46 1999
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From: <abbyjenkins@bigfoot.com>
Subject: Become a reviewer for iNet Reviews and receive books, software, products you can keep for free!
Date: Sun, 10 Oct 1999 18:51:05
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iNet News October 1999 (for reviewers and prospects only)

INet Reviews needs book, software & product reviewers. You may 
request any virtually title or consumer product. Titles are 
shipped directly to you with the freight paid from the publisher 
or manufacturer. You write a brief review and post it to our web 
site. 

For your review starter information kit reply to this email or 
send an email message with the words STARTER 1010g53 in the 
message subject line to  juliacantor@bigfoot.com The starter kit 
contains complete information plus a 70 page list of review 
products others have received.

This message is directed for iNet reviewers and referrals only. 
If you received this message by mistake, reply with the single 
word REMOVE to have your name removed from our email message 
database.

If you have problem responding to this email, send a Self 
addressed Stamped envelope to iNet Reviews, 4015 Holcomb bridge 
Road, Ste 350-905, norcross, GA 30092.
 
 
 
 
 
 
 
 



From rem-conf-request@es.net  Mon Oct 11 00:08:21 1999
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Date: Mon, 11 Oct 1999 20:52:28 -0700
To: rem-conf@es.net
From: Ross Finlayson <finlayson@live.com>
Subject: Updated info on tools for receiving MP3/RTP sessions
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FYI, here is some updated information on available tools for receiving and
playing multicast MP3 sessions (several of which have been announced on the
MBone recently).  There have recently been some new tools for receiving
these sessions, as well as improvements to some existing tools.

1/ "Freeamp" <http://www.freeamp.org/> - for Windows and Linux/x86 (&
probably other platforms in the future) - can receive and play MP3/RTP
multicast sessions, by giving it a 'pseudo-URL' argument:
	rtp://<multicast-address>:<port>
(Thanks to Robert Kaye for implementing RTP reception in Freeamp.)  This
implementation currently does not generate RTCP reports.

2/ A new input plugin for AOL/Nullsoft's "Winamp" audio player (for
Windows) allows Winamp users to receive & play MP3/RTP sessions directly.
For details, see <http://www.live.com/multikit/winamp-plugin.html>.

3/ For other MP3 player tools, the "playRTPMPEG" helper application
<http://www.live.com/multikit/playRTPMPEG.html> can be used.  This tool
receives multicast MP3/RTP data, and feeds the MP3 data to an existing
audio player (either via stdout, or via a built-in HTTP server).

In particular, RealNetworks' "RealPlayer G2" can now be used - in
combination with "playRTPMPEG" - to play multicast MP3 streams.

For details, see <http://www.live.com/multikit/playRTPMPEG.html>

Both "playRTPMPEG" and the new Winamp input plugin can now support the
experimental "X-MP3" alternative RTP payload format for MP3 data, as well
as the conventional MPEG audio format ("MPA") described in RFC2250.  (The
"liveCaster" server tool can also (optionally) transmit MP3 data using this
new payload format.)  The new payload format is more tolerant of network
packet loss.  For more information, see <http://www.live.com/rtp-mp3/>.  (I
will also be updating and resubmitting the associated Internet-Draft prior
to the Washington DC IETF meeting.)

	Ross.





From rem-conf-request@es.net  Mon Oct 11 10:07:31 1999
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Date: Mon, 11 Oct 1999 08:38:44 -0500
From: Ken Hopper <k-hopper@uchicago.edu>
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REMOVE



From rem-conf-request@es.net  Tue Oct 12 03:02:56 1999
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	Tue, 12 Oct 1999 08:26:39 +0200 (MET DST)
From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "Larry Masinter" <masinter@parc.xerox.com>,
        "Harald Tveit Alvestrand" <Harald@Alvestrand.no>,
        <ietf-types@iana.org>, "IETF-avt list" <rem-conf@es.net>
Cc: "Thomas Johansson" <thomas.johansson@omnitor.se>
Subject: SV: Registration of MIME media type text/t140
Date: Tue, 12 Oct 1999 08:25:16 +0200
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Larry Masinter asks if the Text Conversation topic should be handled by the
IMPP group.
Yes, why not.
Now, the transport is handled in the avt group because it makes use of RTP.
Then the t140 medium is to be registered as a MIME text type.

The purpose is real time character by character ( or close ) transmission of
text in a call.
In many cases it is combined with voice or video.

IMPP goals is short messages first entered as a whole and then transmitted.
That is different, but it would of course be of interest to study
interworking.

Regards
Gunnar Hellström


> -----Ursprungligt meddelande-----
> Från: Larry Masinter [mailto:masinter@parc.xerox.com]
> Skickat: den 11 oktober 1999 01:53
> Till: gunnar.hellstrom@omnitor.se; Harald Tveit Alvestrand;
> ietf-types@iana.org
> Ämne: RE: Registration of MIME media type text/t140
>
>
>
> > Response to Harald's questions:
> >
> > The MIME Media type text/t140 is intended for the standards track.
> > It will be used firstly in the megaco/H.248 protocol to indicate use of
> > either RTP/t140 with the RTP payload format specified in the
> internet-draft
> > provided as the transport and medium for text conversation, or
> TCP with the
> > t140 stream without extra framing.
> >
> > For that purpose it also needs IANA registration.
> >
> > T.140 is intended for real time, character by character text
> chatting. Time
> > stamping info is not important. In transmission, it is UTF-8
> coded. The use
> > as a text chat medium restricts use of positioning and
> formatting control
> > characters to a minimal set.
> >
> > To me it seems like it is suitable to register it as text.
> >
> > There is no explicit way to indicate language.
> > It may be hard in an real time text chat to promise to keep to
> one language,
> > but there are many other planned environments where T.140 fits,
> so it might
> > be useful to give a language indication.
> >
> > Can you advice me how that is normally done?
> >
>
> The IMPP working group is developing standards for real-time
> text chatting. If there's going to be an IETF standard for
> it, shouldn't it be considered by the IMPP group?
>
>
----------------------------------------------
Gunnar Hellström
LM Ericsson
Sweden
E-mail gunnar.hellstrom@omnitor.se

Tel +46 8 556 002 03
Mob. +46 708 204 288
Video and V.18 text tel +46 8 556 002 05
Fax +46 8 556 002 06
------------------------------------------




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Subject: I-D ACTION:draft-ietf-avt-rtp-text-01.txt
Date: Tue, 12 Oct 1999 06:55:26 -0400
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload for Text Conversation
	Author(s)	: G. Hellstrom
	Filename	: draft-ietf-avt-rtp-text-01.txt
	Pages		: 7
	Date		: 11-Oct-99
	
This memo describes how to carry text conversation session contents
in RTP packets. Text conversation session contents is specified in
ITU-T Recommendation T.140 [1]. 
Text conversation is used alone or in connection to other
conversational facilities such as video and voice, to form multimedia
conversation services.
This RTP payload description contains an optional possibility to
include redundant text from already transmitted packets in order to
reduce the risk of text loss caused by packet loss. The redundancy
coding follows RFC 2198.

A URL for this Internet-Draft is:
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From rem-conf-request@es.net  Tue Oct 12 09:56:05 1999
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To: rem-conf@es.net
Subject: Re: WG last call: MIB and tones drafts
In-reply-to: Your message of "Sun, 03 Oct 1999 17:12:11 BST." <4175.938967131@cs.ucl.ac.uk>
Date: Tue, 12 Oct 1999 14:38:23 +0100
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From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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A reminder that this working group last call is in progress. If you have
any final comments on these drafts, please send them by Monday.

Thanks,
Colin


--> Colin Perkins writes:
>To the AVT working group,
>
>This is to announce a two week working group last call on the following
>drafts:
>
>Title     : RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
>Author(s) : H. Schulzrinne, S. Petrack
>Filename  : draft-ietf-avt-tones-01.txt
>Pages     : 25
>Date      : 27-Sep-99
>
>Title     : Real-Time Transport Protocol Management Information Base
>Author(s) : M. Baugher, I.  Suconick,  B. Strahm
>Filename  : draft-ietf-avt-rtp-mib-06.txt
>Pages     : 37
>Date      : 28-Sep-99
>
>The DTMF draft addresses the concerns raised during the Oslo meeting, and
>the MIB addresses comments raised during the previous working group last
>call held earlier this year.
>
>Please send comments to the rem-conf@es.net list before 18th October. If
>there are no serious objections, we intend to request the IESG to consider
>these for publication as proposed standards at that time.
>
>Colin
>



From rem-conf-request@es.net  Tue Oct 12 17:23:37 1999
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From: Andrew Daviel <andrew@andrew.triumf.ca>
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Subject: pruning required in mrouted ?
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We are basically a leaf node on the Mbone. We had been running
mrouted 3.8 on an SGI but it has been a bit unreliable, and I am
considering using our FORE PowerHub 7100 directly.

The PowerHub is an older model; according to mrinfo it is running "version
2.0". Thus, I believe it is not running prune and is not trace-capable. A
software upgrade may be impossible without replacing some hardware.

Is pruning absolutely required everwhere, or just at routing nodes?
I haven't been following the list for a while; I remember some push to
make sure no-one ran mrouted3.7.

Would it cause trouble if we enable multicast on this router ?

Andrew Daviel
TRIUMF




From rem-conf-request@es.net  Wed Oct 13 09:25:34 1999
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From rem-conf-request@es.net  Wed Oct 13 13:39:18 1999
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From: almeroth@cs.ucsb.edu (Kevin C. Almeroth)
Message-Id: <199910131717.KAA21862@jackson.cs.ucsb.edu>
To: brutzman@nps.navy.mil, rem-conf@es.net
Subject: Re:  rtpMonitor thesis and software available
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>>The thesis and software are available at
>>http://www.web3D.org/WorkingGroups/vrtp/rtpMonitor/rtpMonitor.html
>>Sample screen snapshots are available at
>>http://www.web3D.org/WorkingGroups/vrtp/rtpMonitor/manual.html

Not that this tool is available (and I actually got it to work on
Windoze) can anyone advise on how to make .vic.tcl and .vat.tcl
work in Windoze?  Or, does it simply not work?

I'd like to add the rtpmon button to my windows apps.

-Kevin



From rem-conf-request@es.net  Wed Oct 13 13:48:21 1999
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CC: brutzman@nps.navy.mil, rem-conf@es.net
Subject: Re: rtpMonitor thesis and software available
References: <199910131717.KAA21862@jackson.cs.ucsb.edu>
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Hi Kevin;

I seem to remember seeing that the call to source the .vic.tcl 
was commented out in some versions of the source.

(I was looking for it on an older of vic for Linux)

Do you have the newest vic source from UCL? Maybe if you don't
get any informed answers you can have a peek at the
source code to see what happened to the source line.
Just a thought...

John Stewart
john.stewart@crc.ca


> Not that this tool is available (and I actually got it to work on
> Windoze) can anyone advise on how to make .vic.tcl and .vat.tcl
> work in Windoze?  Or, does it simply not work?
> 
> I'd like to add the rtpmon button to my windows apps.



From rem-conf-request@es.net  Thu Oct 14 10:35:17 1999
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Date: Thu, 14 Oct 1999 06:57:13 -0700 (PDT)
From: almeroth@cs.ucsb.edu (Kevin C. Almeroth)
Message-Id: <199910141357.GAA24266@jackson.cs.ucsb.edu>
To: rem-conf@es.net
Subject: Re:  rtpMonitor thesis and software available
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>>Not that this tool is available (and I actually got it to work on
>>Windoze) can anyone advise on how to make .vic.tcl and .vat.tcl
>>work in Windoze?  Or, does it simply not work?

Thanks to Marcia Perry at LBL I'm part of the way there.
I can get the button to show up (.vic.tcl and .vat.tcl need
to be in c:\ not mbone home).

But now I can't get the exec to exec ANYTHING...  I've even
tried simple bat files stored all over the place, commands
like "dir" and "set".  I'm still hunting though.

-Kevin



From rem-conf-request@es.net  Thu Oct 14 12:05:59 1999
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Date: Thu, 14 Oct 1999 17:48:57 +0200 (MET DST)
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To: "Kevin C. Almeroth" <almeroth@cs.ucsb.edu>
cc: rem-conf@es.net
Subject: Re: rtpMonitor thesis and software available
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On Thu, 14 Oct 1999, Kevin C. Almeroth wrote:

> >>Not that this tool is available (and I actually got it to work on
> >>Windoze) can anyone advise on how to make .vic.tcl and .vat.tcl
> >>work in Windoze?  Or, does it simply not work?
> 
> Thanks to Marcia Perry at LBL I'm part of the way there.
> I can get the button to show up (.vic.tcl and .vat.tcl need
> to be in c:\ not mbone home).
> 
> But now I can't get the exec to exec ANYTHING...  I've even
> tried simple bat files stored all over the place, commands
> like "dir" and "set".  I'm still hunting though.

The sdr (maybe depends on the tcl/tk for Windows) would exec only *.exe files.
Try to use "exec start command /c xxx.bat &" on win9x or 
"exec cmd /C start xxx.bat &" on NT.

You can find a example tcl file (start ReLaTe from sdr):
ftp://ftp-mm.urz.tu-dresden.de/pub/mbone/relate/mmrz/sdr-relate.tcl

Rgds,
Christoph Fleck

,----------------------------------------------------------------.
| Beratungszentrum fuer Videokonferenzdienste (BZVD), TU Dresden |
| Christoph Fleck                      Tel.: +49 (0)351/463-5653 |
| E-Mail: bzvd@tu-dresden.de       http://bzvd.urz.tu-dresden.de |
`----------------------------------------------------------------'




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Date: Thu, 14 Oct 1999 09:30:00 -0700
To: Florissa Colina <florissa@gumby.CS.Berkeley.EDU>
From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 10/20 New Approaches to Image Retrieval and Video Shot
  Boundary Detection -- Yihong Gong, NEC 
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Berkeley Multimedia, Graphics and Interfaces Seminar

New Approaches to Image Retrieval and Video Shot Boundary Detection

Wednesday October 20, 1999, 
1:00-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 


Yihong Gong 
NEC

The explosive growth of digital images in computer systems and networks has
dramatically increased the
need for multimedia database systems that effectively index, store, and
retrieve image and video data based
on their contents. Techniques for content-based image retrieval enable
users to retrieve images based on their
visual similarities, while techniques for video shot boundary detection aim
to segment a continuous video
sequence into visually consistant units, so that the sequence can be
efficiently indexed and retrieved. In this
talk, I will present two novel approaches for image retrieval and video
shot boundary detection. 

First, I will present an image retrieval system that strives to achieve
advanced content-based image retrieval
using seamless combination of two complementary approaches: on one hand, we
propose a new color
clustering method to better capture color properties of the original
images; on the other hand, expecting that
image regions acquired from the original images inevitably contain many
errors, we make use of the available
erroneous, ill-segmented image regions to accomplish the object
region-based image retrieval. We also
propose an effective image indexing scheme to facilitate fast and efficient
image matching and retrieval. 

Next, I will present a new shot boundary detection method that achieves a
better shot boundary detection rate
and the precision rate. We use the same color clustering method used in the
image retrieval system for
extracting color distributions from video frames. A novel approach is
proposed to effectively suppress both the
intra-frame and the inter-frame color variations caused by noise, minor
illumination changes, as well as
camera and object motions.

---------------

The seminar is broadcast live on the Internet Mbone and as a Real
Networks G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player:

rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
   http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving
the announcement you may be able to receive the session by manually
configuring the client programs ('vic', and 'vat') with the session
addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to
replays of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




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From: "Peter Lewis" <peter.lewis@upperside.fr>
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Subject: Media Gateway Control
Date: Fri, 15 Oct 1999 14:27:05 +0200
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MGCP, Megaco, H.248: performing seamless interoperation between IP and =
PSTN
networks. The international rendez-vous in Paris, 15-17 December 1999.

More infos:=20
http://www.upperside.fr/bamgc.htm


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<DIV><FONT size=3D3>MGCP, Megaco, H.248: performing seamless =
interoperation=20
between IP and PSTN<BR>networks. The international rendez-vous in Paris, =
15-17=20
December 1999.<BR><BR>More infos: </FONT></DIV>
<DIV><FONT color=3D#800080 size=3D3><A=20
href=3D"http://www.upperside.fr/bamgc.htm">http://www.upperside.fr/bamgc.=
htm</A></FONT></DIV></DIV>
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From rem-conf-request@es.net  Fri Oct 15 08:59:53 1999
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From: The IESG <iesg-secretary@ietf.org>
Subject: Protocol Action: An RTP Payload Format for Generic Forward
	 Error Correction to Proposed Standard
Date: Fri, 15 Oct 1999 08:33:47 -0400
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The IESG has approved the Internet-Draft 'An RTP Payload Format for
Generic Forward Error Correction' <draft-ietf-avt-fec-08.txt> as a
Proposed Standard.  This document is the product of the Audio/Video
Transport Working Group.  The IESG contact persons are Scott Bradner
and Vern Paxson.


Technical Summary
 
The document provides a means of providing FEC for an RTP flow by XOR'ing
together data packets to generate a separate FEC stream.  The FEC format is
generic in the sense that it works with arbitrary algorithms for deciding
which data packets to combine and how many FEC packets to generate.  There
are two ways in which the FEC data may be transported: as a separate RTP
flow, or by piggy-backing onto the original flow using the RFC2198 "redundant
audio" payload format.

Working Group Summary

The FEC process was not controversial in AVT, and the draft includes
appropriate warnings about misuse.  The draft was held up in the working
group for some time because it proved difficult to define SDP parameters to
describe transport of FEC as a separate flow.  A clean solution would require
grouping the FEC flow with the original media flow, but SDP doesn't support
grouping. The solution adopted in the draft is to include an SDP connection
specification as a codec parameter, indicating where the FEC information for
a flow is to be found.  This is ugly, but the only workable solution the
WG found.

Protocol Quality

The document was reviewed for the IESG by Vern Paxson.




From rem-conf-request@es.net  Fri Oct 15 11:47:43 1999
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To: rem-conf@es.net
Subject: AVT agenda and scheduling
Date: Fri, 15 Oct 1999 16:27:09 +0100
Message-ID: <3079.940001229@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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The AVT working group has two slots scheduled at the Washington DC IETF
meeting:
	Wed 10th November, 15:30-17:30
	Thu 11th November, 09:00-10:30
Those wishing agenda time in those slots should contact myself or Steve,
and we'll put together a draft agenda.

Thanks,
Colin



From rem-conf-request@es.net  Fri Oct 15 20:51:49 1999
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From: "ray@netondemand.com" <ray@netondemand.com>
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Subject: FREE ISP DISCONNECT IDLE SOLUTION
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ELIMINATE THIS PROBLEM FOR GOOD:

"YOU HAVE BEEN IDLE FOR XX MINUTES, YOU WILL BE 
DISCONNECTED..."
 
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From rem-conf-request@es.net  Sat Oct 16 10:28:49 1999
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Date: Sat, 16 Oct 1999 10:00:17 -0400 (EDT)
From: Jaiwant Mulik <jmulik@astro.ocis.temple.edu>
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To: rem-conf@es.net
Subject: Calculating Control messages bandwidth.
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Hi, 

The SRM famework suggests that all control traffic be limited to 5% of the
total traffic. How can I estimate the total traffic so far and keep my
control messages limited to within the 5% constraint? Is the
limit on a per host basis or for the whole group? Pointers to relevent
literature would be highly appreciated.

regards,
-----------------------------------------------------------------------
Jaiwant Mulik                             Home : (215) 204-5775
Room 203, 1928 N Broad St.                Email: jmulik@unix.temple.edu
Philadelphia, PA                               : jmulik@hotmail.com
USA 19121                                      : jmulik@acm.org
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Lekin woh zindagi he kya, jisme koi namumkin sapna na ho ?




From rem-conf-request@es.net  Tue Oct 19 07:20:11 1999
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload Format for DV Format Video
	Author(s)	: K. Kobayashi, A. Ogawa,  S. Casner, C. Bormann
	Filename	: draft-ietf-avt-dv-video-01.txt
	Pages		: 11
	Date		: 18-Oct-99
	
This document specifies the packetization scheme for encapsulating DV
compressed digital video data streams, commonly known as

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-dv-video-01.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: Guidelines for Writers of RTP Payload Format 
                          Specifications
	Author(s)	: M. Handley, C. Perkins
	Filename	: draft-ietf-avt-rtp-format-guidelines-04.txt
	Pages		: 9
	Date		: 18-Oct-99
	
This document provides general guidelines aimed at assisting the authors
of RTP [9] Payload Format specifications in deciding on good formats.
These guidelines attempt to capture some of the experience gained with
RTP as it evolved during its development.
The principles outlined in this document are applicable to almost all
data types, but are framed in examples of audio and video codecs for
clarity.

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From rem-conf-request@es.net  Tue Oct 19 13:44:45 1999
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Date: Tue, 19 Oct 1999 10:23:08 -0700
To: Florissa Colina <florissa@gumby.CS.Berkeley.EDU>
From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 10/20 New Approaches to Image Retrieval and Video Shot
  Boundary Detection -- Yihong Gong, NEC 
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Berkeley Multimedia, Graphics and Interfaces Seminar

New Approaches to Image Retrieval and Video Shot Boundary Detection

Wednesday October 20, 1999, 
1:00-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 


Yihong Gong 
NEC

The explosive growth of digital images in computer systems and networks has
dramatically increased the
need for multimedia database systems that effectively index, store, and
retrieve image and video data based
on their contents. Techniques for content-based image retrieval enable
users to retrieve images based on their
visual similarities, while techniques for video shot boundary detection aim
to segment a continuous video
sequence into visually consistant units, so that the sequence can be
efficiently indexed and retrieved. In this
talk, I will present two novel approaches for image retrieval and video
shot boundary detection. 

First, I will present an image retrieval system that strives to achieve
advanced content-based image retrieval
using seamless combination of two complementary approaches: on one hand, we
propose a new color
clustering method to better capture color properties of the original
images; on the other hand, expecting that
image regions acquired from the original images inevitably contain many
errors, we make use of the available
erroneous, ill-segmented image regions to accomplish the object
region-based image retrieval. We also
propose an effective image indexing scheme to facilitate fast and efficient
image matching and retrieval. 

Next, I will present a new shot boundary detection method that achieves a
better shot boundary detection rate
and the precision rate. We use the same color clustering method used in the
image retrieval system for
extracting color distributions from video frames. A novel approach is
proposed to effectively suppress both the
intra-frame and the inter-frame color variations caused by noise, minor
illumination changes, as well as
camera and object motions.

---------------

The seminar is broadcast live on the Internet Mbone and as a Real
Networks G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player:

rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
   http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving
the announcement you may be able to receive the session by manually
configuring the client programs ('vic', and 'vat') with the session
addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to
replays of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf-request@es.net  Tue Oct 19 23:58:04 1999
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Date: Tue, 19 Oct 1999 20:33:23 -0700
To: rem-conf@es.net
From: Ross Finlayson <finlayson@live.com>
Subject: Updated MP3/RTP Internet Draft submitted
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FYI, I have submitted a updated version of the "A More Loss-Tolerant RTP
Payload Format for MP3 Audio" Internet Draft.  The major changes
(additions) to the draft are:
- a brief section describing how to deal with (unusual) streams that mix
layer I and II audio frames with layer III frames.
- an appendix with 'pseudo code' that describes how to convert a stream of
regular MP3 frames into "ADU frames", or vice-versa.

Until the draft is announced (likely as <draft-ietf-avt-rtp-mp3-01.txt>),
you can find it online at <http://www.live.com/rtp-mp3/rtp-mp3.txt>.  (For
an illustration - with MP3 files - of the benefits of the new format, see
<http://www.live.com/rtp-mp3/>)

I'm interested in any feedback on this draft, especially from MP3 experts
(e.g., at Fraunhofer), or from others who are working on implementations. 

	Ross.

Abstract

While the RTP payload format defined in RFC 2250 is generally applicable
to all forms of MPEG audio or video, it is less suitable for MPEG 1 or 2,
layer III audio (commonly known as "MP3").  The reason for this is that
an MP3 frame is not a true "Application Data Unit" - it contains a
back-pointer to data in earlier frames, and so cannot be decoded
independently of these earlier frames.  Because RFC 2250 defines that
packet boundaries coincide with frame boundaries, it handles packet loss
inefficiently when carrying MP3 data.  The loss of an MP3 frame will
render some data in previous (or future) frames useless, even if they are
received without loss.

In this document we define a new RTP payload format for MP3 audio.
The new format is essentially the same as that defined in RFC 2250, except
for a data-preserving rearrangement of the original MPEG frames, so that
packet boundaries now coincide with true MP3 "Application Data Units".
This new format is therefore more data efficient in the face of packet
loss.





From rem-conf-request@es.net  Wed Oct 20 06:42:36 1999
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To: mbone@ISI.EDU, rem-conf@es.net, release@cs.ucl.ac.uk
cc: P.OHanlon@cs.ucl.ac.uk
Subject: New UCL vic release (vic-2.8ucl-1.0)
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Date: Wed, 20 Oct 1999 11:19:59 +0100
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From: "Piers O'Hanlon" <P.OHanlon@cs.ucl.ac.uk>
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Hi folks,

We are proud to release the new version vic-2.8ucl-1.0.

Vic was developed by LBL as a video conferencing tool to facilitate video 
conferencing over the Internet. This can be between two participants directly, 
or between a group of participants on a common multicast group. No special 
features are required to use VIC in point to point mode, but to use the 
multiparty conferencing facilities of VIC, all participants must reside on a 
portion of the Internet which supports IP multicast. VIC is based on IETF 
standards, using RTP above UDP/IP as its transport protocol, and conforming to 
the RTP profile for audio and video conference with minimal control.

Vic provides a range of video codecs (H.261 Intra, H.263, H.263+, jpeg, bvc, 
cellb, nv, nvdct) from a number of different contributers.
It runs on a range of platforms: Solaris, IRIX, Linux, FreeBSD3.2 and Windows 
95/98/NT.

At UCL we have continued the development of vic (from LBL) and are releasing a 
new version (vic2.8ucl-1.0)

With the following additions, since vic2.8ucl5 (an experimental release):
- Reorganised code into directory structure reflecting the functionality of
  the different modules of vic - in a style not disimilar to MASH. 
- Moved vic over to use UCL's version's of tcl/tk - this is due to a number of 
   problems we have encountered here; The variety of versions of tcl/tk
  installed on target machines can be very wide and therefore provides us with 
  a multitude of bugs and 'problems' that are not related to our code. We
  now ship the code now with UCL statically linkable version of tcl/tk (which  
  also uses compiled in versions of the 'init' scripts as opposed to system
  installed which also lead to trouble) to minimise such problems. It is still 
  possible to get vic to use the standard libraries by specifying 
  -without-ucltk, and/or -without-ucltcl (though we cannot offer much help 
with
  problems created by different tcl/tk versions). 
- Moved vic over to use UCL's common library so it shares the codebase 
provided   by rat - where possible; It now uses common DES code and
  Mbus code - so both the DES and Mbus sections have been re-written.
- The meteor grabber has been fixed (using the source from the freeBSD 
version)
  for FreeBSD and vic now compiles under FreeBSD. 
- A number of useful fixes for the video4linux grabber plus new control panel 
  from Jean-Marc Orliaguet 
- IPv6 support under UNIX and WIN32. Under UNIX this enabled by specifiying 
the
  --enable-ipv6 directive to configure. We have tested it with
  Solaris 7 ipv6, Linux 2.2.5 ipv6 kernel support. 
- Created an MS Visual C++ 6.0 project for compilation of vic under win32 
  environments - this also, unfortunately, meant we had to rename all C++
  files from .cc to .cpp. The project has a number of different configuration  
  choices; Debug, Release, Debug_IPv6 (Microsoft research's IPv6 -
  requiring NT 4.0, MS ipv6 version 1.3 and MS DDK), Debug_IPv6-Musica (a  
  non-free ipv6 implementation from Thompson-CSF). 
- Now IPv6 host addresses and multicast addresses are automatically recognised 
  and vic will attempt to use IPv6 (if compiled in). 

We look forward to any feedback, bug reports, patches and comments on
experience. 

More information, including download instructions, is available from
http://www-mice.cs.ucl.ac.uk/multimedia/software

Please send all feedback to vic@cs.ucl.ac.uk

Piers O'Hanlon/Kris Hasler
Networked Multimedia Group
University College London





From rem-conf-request@es.net  Wed Oct 20 15:27:46 1999
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Date: Wed, 20 Oct 1999 11:48:39 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: ietf-types@iana.org
Subject: Registration of RTP payload format names
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[bcc'd to rem-conf for the AVT working group]

This is a request from the Audio/Video Transport working group to
review draft-ietf-avt-rtp-mime which defines a procedure to register
Real-time Transport Protocol payload formats (encodings) as audio or
video MIME subtype names.  The draft also registers all the RTP
payload formats defined in the RTP Profile for Audio and Video
Conferences as MIME subtypes (RFC 1890 which is being revised as
draft-ietf-avt-profile-new).  The move to establish this procedure was
instigated by Keith Moore and Harald Alvestrand for two purposes:

  - To use the existing MIME namespace for audio and video encodings
    to be transported by RTP rather than creating a new namespace as
    RFC 1890 implicitly did.

  - To increase the number of encodings available for use with
    existing MIME transport modes.

Not all of the RTP payload formats represent encodings that make sense
for other transport modes, or at least it will be necessary to specify
how the media data should be packaged for those modes since just
concatenating a sequence of packet payloads won't be sufficient.  The
draft currently just registers the names for RTP transport and leaves
the specification of packaging for other modes to future drafts.

Your review is requested in particular for the registration procedure
which says what additional information is required to specify RTP
transport of a media type.  The current version of the draft is
draft-ietf-avt-rtp-mime-00.txt, but it is about to be updated to -01
(before draft deadline) with some minor additions.  If anyone has
comments right away, those can be addressed in the update, otherwise
the review will be on the -01 version.

As new RTP payload formats are defined in the future (beyond those
included in RFC 1890 and its revision), each of those RFCs should
include its own registration template.

I believe you have already been informed of one such RTP payload
format specification that happens to fall under "text" rather than
"audio" or "video", i.e. draft-ietf-avt-rtp-text-XX.

							-- Steve




From rem-conf-request@es.net  Wed Oct 20 16:28:19 1999
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Date: Wed, 20 Oct 1999 13:14:45 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: rem-conf@es.net
Subject: RTP spec questions
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AVT working group:

I have a few questions regarding changes to the RTP spec.  I'm
interested in any comments you may have now, and we can discuss these
at the upcoming meeting as well.

1. At the Oslo IETF meeting, and as reported in the minutes of that
meeting, we agreed to change the representation of the loop/collision
detection algorithm to pseudo-C.  This was part of the question, when
a new source transport address is seen for a known SSRC id, whether
packets from the old address should be kept and from the new address
ignored, as is currently specified, or vice versa.  The motivation for
changing the spec is to accommodate mobile hosts that may change their
address.  Since following the new source requires hysteresis in case
there really are two sources colliding, I propose to not add any
additional complexity to the pseudo-code algorithm, but to add in the
text the following statement:

    For applications in which sources may change addresses, the RTP
    implementation SHOULD modify the collision detection algorithm to
    accept packets from the new source transport address.  To guard
    against flip-flopping between addresses if a genuine collision
    does occur, the algorithm should include some means to detect this
    case and avoid switching.

2. As discussed in August, if a source has not been sending data for a
while, so it is sending RR's with a fairly large interval according to
the receiver fraction of the RTCP bandwidth, and then it begins to
transmit, it may be quite a while before the RTCP interval expires and
the first SR is sent.  The change is to add a statement to 6.3.8 that
a transition of we_sent from false to true SHOULD trigger a "reverse
reconsideration" recalculation of the RTCP interval.

3. In scenarios with one or a small number of data senders and many
receivers, the sender's RTCP interval may be much shorter than that of
the receivers.  Therefore, 6.3.5 on timing out an SSRC needs to be
based on the RTCP interval using the receiver fraction rather than the
sender fraction, even for senders.

4. In Oslo we decided _not_ to change the calculation of avg_rtcp_size
to include only the RTCP packets being transmitted and exclude those
received from other sources, even though there is a potential for
unfair use of RTCP bandwidth, because it would complicate many parts
of the algorithm without a clear gain.

A related question has occurred to me just recently:  If an RTCP
compound packet includes information from more than one host, as in
the case where an RTP mixer aggregates SDES information from multiple
sources, should the size of that packet be divided by the number of
sources before being figured into the average?

The scenario which tiggered this question was the idea we discussed in
Oslo for a possible new one-to-many RTP profile in which RTCP packets
are sent via unicast from receivers to the sender which then reflects
them back out to all receivers using multicast.  Packets from multiple
receivers could be aggregated, which would perturb the existing
average size calculation.  For a new profile, it would be possible to
specify a modification to the calculation for that profile.

							-- Steve




From rem-conf-request@es.net  Thu Oct 21 06:55:33 1999
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From: Rafiq Shaikh <rafiq@chiplogic.com>
To: rem-conf@es.net
cc: hariv@chiplogic.com
Subject: Silence Supression - Frame Based Speech Codecs? 
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Hi all,

As far as RTP is concerened, how is Silence Supression implemented w.r.t.
Frame Based speech codecs?

During Silence supression, are RTP packets sent without any Payload or
we don't need to send RTP packets at all?

If No RTP packet is sent for the Silence period:
 a) How do we distinguish a Packet loss with No packet sent? 
   (During Silence frames)
 b) In RTP, is there a restriction on the duration of Silence period?
 c) What is the possible effect on the statistical calculations done in
    the RTCP part.

How can we utilize the Marker bit in the RTP Header to distinguish 
between the NULL frame and the Active frame within the same RTP Packet?

Thanks in advance.

Regards,
-Rafiq.




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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: A More Loss-Tolerant RTP Payload Format for MP3 Audio
	Author(s)	: R. Finlayson
	Filename	: draft-ietf-avt-rtp-mp3-00.txt
	Pages		: 7
	Date		: 20-Oct-99
	
While the RTP payload format defined in RFC 2250 is generally applicable
to all forms of MPEG audio or video, it is less suitable for MPEG 1 or 2,layer III audio (commonly known as 'MP3').  The reason for this is that an MP3 frame is not a true 'Application Data Unit' - it contains a
back-pointer to data in earlier frames, and so cannot be decoded
independently of these earlier frames.  Because RFC 2250 defines that
packet boundaries coincide with frame boundaries, it handles packet loss
inefficiently when carrying MP3 data.  The loss of an MP3 frame will
render some data in previous (or future) frames useless, even if they are received without loss.
In this document we define a new RTP payload format for MP3 audio.
The new format is essentially the same as that defined in RFC 2250, except for a data-preserving rearrangement of the original MPEG frames, so that packet boundaries now coincide with true MP3 'Application Data Units'. This new format is therefore more data efficient in the face of packet
loss.

A URL for this Internet-Draft is:
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From rem-conf-request@es.net  Thu Oct 21 09:58:01 1999
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Subject: RTCP payload format for backward messages
Date: Thu, 21 Oct 1999 22:41:03 +0900
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Dear AVT experts,

Japanese MPEG-4 experts are planning to propose a new RTP format for 
MPEG-4 Audio/Visual to Washington meeting. This proposal was discussed 
in the last MPEG meeting (October), and ISO send a liaison to IETF to 
recommend this new proposal. Now we are discussing this issue on the 
MPEG-4 over IP AdHoc Group's reflector. We will input the internet-draft 
soon until tomorrow dead line.

In this internet-draft, there is a new proposal of RTCP payload format, 
as well as RTP formats.

This RTCP payload will be used to transport the backward messages of 
NEWPRED from the decoder to the encoder. 
NEWPRED is an error resilience video tool selecting the reference pictures 
according to the network error conditions. 

The syntax of the backward messages of NEWPRED is defined in the Visual 
part of MPEG-4 version 2, and the streams of the back messages are 
generated in the video decoder corresponding to the forward Video Packets 
or VOPs. The contents of the messages are changed message by message.
Therefore the RTCP, we proposed, has a payload to deliver the backward 
message as below.

----
     0                   1                   2                   3
     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |V=2|P|   BMT   |  $B!!!!!!(BPT $B!!!!!!(B|           length              |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |                              SSRC                             |
    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
    |    MPEG-4 Backward Channel Message Payload (byte aligned)     |
    |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |                               :             padding           |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

version (V): 2 bits
Identifies the version of RTP, which is the same in RTCP packets
as in RTP data packets. 

padding (P): 1 bit
If the padding bit is set, this RTCP packet contains some
additional padding octets at the end which are not part of the
control information. The last octet of the padding is a count of
how many padding octets should be ignored. Padding may be needed
by some encryption algorithms with fixed block sizes. In a
compound RTCP packet, padding should only be required on the
last individual packet because the compound packet is encrypted
as a whole.

backward channel message type (BMT): 5 bits
 Identifies the type of the MPEG-4 backward channel messages.
 0:  forbidden
 1:  MPEG-4 Visual NEWPRED
 2-63: reserved
 In this internet-draft, only NEWPRED is assigned as the candidate
 of the BMT for the moment.  Some other MPEG-4 Audio/Visual applications using
 the backward channel messages may be assigned in the future. 

packet type (PT): 8 bits
 The value of the packet type (PT) identifier is the constant 
 for MPEG-4 backward channel messages. The assignment of an RTCP payload type for this new packet format is outside the scope of 
this document, and will not be specified here.

SSRC: 32 bits
 SSRC is the synchronization source identifier for the sender of 
 this packet. 

MPEG-4 Backward Channel Message Payload: variable
 The syntax and semantics of the MPEG-4 backward channel messages 
 are defined in the ISO/IEC 14496-2/3. All messages are
 byte aligned.  Normally one message is mapped onto one RTCP 
 packet, and several messages with same BMT could be continuously mapped onto one
 RTCP packet.  One message SHALL NOT be fragmented into different 
RTCP packets.
--


To tell the truth, we had wondered which format (RTP or RTCP) is 
appropriate for the backward messages of NEWPRED, and we decided to use 
RTCP at the Melbourne. If there is other good solution, would you please 
propose it, and discuss here and at Washington?

We understand that RTCP has a limit of sending timing, and the receiver 
has to send RTCP report to all terminals. However there is a exceptional 
procedure in the RFC2032 (RTP payload format for H.261). In the document, 
the control messages can be sent immediately after error detection only to 
one encoder (not all terminals).
Therefore we copy the description of RFC2032 to our proposal for the moment.

Comments are welcome.

Best Regards,
Shigeru Fukunaga
--
   Shigeru Fukunaga (fukunaga444@oki.co.jp)
   Oki Electric Industry Co., LTD.
   Tel. +81 6 6949 5105; Fax. +81 6 6949 5108






From rem-conf-request@es.net  Thu Oct 21 11:01:05 1999
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Subject: Re: Silence Supression - Frame Based Speech Codecs?
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As far as RTP is concerened, how is Silence Supression implemented w.r.t.
Frame Based speech codecs?

During Silence supression, are RTP packets sent without any Payload or
we don't need to send RTP packets at all?
Answer:  I believe you do not need to send any packets at all. This is
vocoder specific, however, all of
the ones which I am familiar with send special packets at the beginning of
a silence period and periodically
during one (at a very reduced rate).

If No RTP packet is sent for the Silence period:
 a) How do we distinguish a Packet loss with No packet sent?
   (During Silence frames)
Answer:  Excellent question.  I believe that you cannot tell.  There are
some augmentations in the works for
RTP such as redundancy which attempt to grapple with this.
 b) In RTP, is there a restriction on the duration of Silence period?
Answer:  As far as I know, RTP itself imposes no restriction.
 c) What is the possible effect on the statistical calculations done in
    the RTCP part.
Answer:  I believe there is no major effect on the calculation of jitter.
Packet loss however does have
an effect.

How can we utilize the Marker bit in the RTP Header to distinguish
between the NULL frame and the Active frame within the same RTP Packet?
Answer:  Again this is vocoder specific.  Some vocoders have rules about
what sort of frame can follow another frame.  Others have implicit length
indicators within each frame.  The marker bit is essentially
useless/redundant in my humble opinion.

Hope that helps.
Other replies invited.





From rem-conf-request@es.net  Thu Oct 21 15:38:14 1999
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From: Alexander Tulai - icpd <alexander_tulai@Mitel.COM>
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Dear Colleagues,

I am afraid that proper means for adopting standards are under serious
preasure in the case of a new proposal for CNG coming from Lucent.

I would like to point out to you, that the "urgent work" is actually
urgent due to the presentation of one contribution only, from Lucent,
to the  following ITU working groups:
SG 16 Q19
SG 15, Q6,
SG 15, Q21
These working groups are perhaps not the only ones the contribution went to.
The fact that Lucent wants it badly doesn't mean the ITU community has
to cut corners.

At this point in time, the Comfort Noise Generation for G.711 method adopted by
IETF for IP networks (see RFC1890) and ITU-T for ATM networks (see I.366.2
Appendix
I2) are identical (one byte representing the power level, with the least
significant 7 bits
carrying the level).

I would like to point out to you that noise spectral parameters are sometimes
byproducts
of VAD algorithms. As such, by standardizing one, automatically, a certain
algorithm
must be used as a VAD.
Both ITU and the ATM forum, as I pointed out to Mr Zopf, have decided in the past
not to standardize VAD algorithms for G.711.

I believe that any work in this area has to be done after a very careful
examination of
the implications of breaking the existing CNG mechanism and proposing a new one.
Due ITU-T standardization procedures have to be in place and, if the decision
is taken that such an alhorithm is required, enough time should be given for
proposals
to be advanced from the interested companies.

Best regards,

Alexander Tulai
Mitel Corporation
Ph: (613) 592-2122
Fax: (613) 592-4784
Email: alexander_tulai@mitel.com


Robert Zopf wrote:

> Dear Colleagues,
>
> I would like to commence the work on defining a Comfort Noise Generation
> scheme for speech codecs without a built-in algorithm.  The proposed steps
> to complete this work as outlined by the Rapporteur of Q.19 include :
>
> (1)     Complete the Performance Requirements and Objectives
> (2)     Draft preliminary test methodology and ask Q.14/12 for comments
> (3)     Complete the Test Methodology
> (4)     Collect candidate algorithms for CNG
> (5)     Performance assessment for the candidates
> (6)     Select a candidate and define the algorithm (Draft Appendix to G.711) .
>
> Because the work is urgently needed by WP2/16 and the IETF, the goal should
> be to have a draft appendix ready for approval at the SG16 meeting in Feb.
> 2000.  The plan is to use  wp3audio as the reflector for discussions.
>
> Please offer any comments or suggestions with respect to this proposed
> procedure.
> I will circulate a revised draft of the Terms of Reference for review and
> comments shortly.
>
> Best regards,
>
> Rob.
>
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> Rob Zopf
> Principal Engineer
> Speech Technology Group
> Lucent Technologies, INS
>
> Phone   : (732) 578-3207
> Fax             : (732) 578-3213
> Email           : zopf@lucent.com
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~




From rem-conf-request@es.net  Thu Oct 21 17:07:59 1999
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Hello there...

On Tue, 12 Oct 1999, Andrew Daviel wrote:

> Is pruning absolutely required everwhere, 

I think the answer is basically "yes", though there might be some
exceptional situations.

I'm not sure what kind of multicast support your router is offering. I
guess we both started this at a time when DVMRP with IP-in-IP tunnels
was the only way, but there's all kinds of different ways nowadays!

> or just at routing nodes?

No, a non-pruning leaf node can do just as much harm, as far as I
know.

> I haven't been following the list for a while; I remember some push to
> make sure no-one ran mrouted3.7.

Even 3.8 is buggy, especially in the face of routing instabilities: NB
those instabilities need not be anywhere close to the v3.8 mrouted -
we found all kinds of leakage until we moved to the so-called
3.9beta3, associated with routing instabilities that, as far as we
could tell, were entirely outside of the UK.

> Would it cause trouble if we enable multicast on this router ?

I recommend keeping a close eye on the multicast traffic levels if you
try it.  It's easiest if you have an SNMP-capable mrouted upstream, so
that you can graph its multicast traffic to check what's going on.  
If not, then use mtrace to investigate the aggregate packet rates.

May I suggest you contact me off the list if you're interested in this
issue, and we can report a summary back to the list.

Disclaimer: I'm not an expert in these matters, but I do know a few
of them, and I run an mrouted myself for the Physics dept.

p.s I get excellent results with a quite elderly PC dedicated to 
running mrouted under FreeBSD.




From rem-conf-request@es.net  Thu Oct 21 17:16:10 1999
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alexander_tulai@Mitel.COM wrote:
> 
> Dear Colleagues,
> 
> I am afraid that proper means for adopting standards are under serious
> preasure in the case of a new proposal for CNG coming from Lucent.
> 
> At this point in time, the Comfort Noise Generation for G.711 method adopted by
> IETF for IP networks (see RFC1890) and ITU-T for ATM networks (see I.366.2
> Appendix
> I2) are identical (one byte representing the power level, with the least
> significant 7 bits
> carrying the level).

	Hi Alex, how are you? Sorry, maybe I've missed something but
	I cannot find the RFC1890 reference - which section or page
	number is it on?
 
> I would like to point out to you that noise spectral parameters are sometimes
> byproducts of VAD algorithms. As such, by standardizing one, automatically, a certain
> algorithm must be used as a VAD.

	You don't necessarily have a DSP resource available for  a G.711
	application, do you? Either for VAD or CNG.

> Both ITU and the ATM forum, as I pointed out to Mr Zopf, have decided in the past
> not to standardize VAD algorithms for G.711.

> I believe that any work in this area has to be done after a very careful
> examination of
> the implications of breaking the existing CNG mechanism and proposing a new one.

	What existing CNG mechanism (in VoIP apps, that is)? Many people just
	play D.C. which sounds terrible, I agree. But work has been done in
	a company not a stone's throw from here that produces reasonable
results.

[\] Robert Wood  mailto:rwood@mosaid.com   <---- note address

The St. Lawrence River - fresh, warm, visible diving



From rem-conf-request@es.net  Thu Oct 21 19:01:40 1999
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Date: Thu, 21 Oct 1999 18:47:07 -0400
To: alexander_tulai@mitel.com, wp3audio@ctd.comsat.com
From: Gerard Aguilar <Gerard.Aguilar@ascend.com>
Subject: Re: Study of CNG
Cc: rem-conf@es.net
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Alexander,

Please see my comments below:

Regards,

Gerard

At 03:05 PM 10/21/1999 -0400, Alexander Tulai - icpd wrote:
>Dear Colleagues,
>
>I am afraid that proper means for adopting standards are under serious
>preasure in the case of a new proposal for CNG coming from Lucent.
>
>I would like to point out to you, that the "urgent work" is actually
>urgent due to the presentation of one contribution only, from Lucent,
>to the  following ITU working groups:
>SG 16 Q19
>SG 15, Q6,
>SG 15, Q21
>These working groups are perhaps not the only ones the contribution went to.
>The fact that Lucent wants it badly doesn't mean the ITU community has
>to cut corners.

Lucent is not asking the ITU to cut corners.  It is simply requesting that
the ITU promptly address the deficiencies associated with the current CNG
scheme as outlined in RFC1890 for running VoIP within H.323 sessions.
The primary problem is that there is no descriptor for shaping the spectrum
of comfort noise.



>At this point in time, the Comfort Noise Generation for G.711 method 
>adopted by
>IETF for IP networks (see RFC1890) and ITU-T for ATM networks (see I.366.2
>Appendix
>I2) are identical (one byte representing the power level, with the least
>significant 7 bits
>carrying the level).
>
>I would like to point out to you that noise spectral parameters are sometimes
>byproducts
>of VAD algorithms. As such, by standardizing one, automatically, a certain
>algorithm
>must be used as a VAD.

The need to utilize the feature set generated by CNG analysis/synthesis by 
any VAD
is simply not true.  Although, it could be desirable depending on the 
feature set produced
by a CNG system, it is in no way a requirement.  Therefore, any design 
decisions made
by the CNG will not impact any proprietary VAD.



>Both ITU and the ATM forum, as I pointed out to Mr Zopf, have decided in 
>the past
>not to standardize VAD algorithms for G.711.
>
>I believe that any work in this area has to be done after a very careful
>examination of
>the implications of breaking the existing CNG mechanism and proposing a 
>new one.

We of course agree with this position and look forward to contributions within
the ITU which satisfy this requirement.



>Due ITU-T standardization procedures have to be in place and, if the decision
>is taken that such an alhorithm is required, enough time should be given for
>proposals
>to be advanced from the interested companies.
>
>Best regards,
>
>Alexander Tulai
>Mitel Corporation
>Ph: (613) 592-2122
>Fax: (613) 592-4784
>Email: alexander_tulai@mitel.com
>
>
>Robert Zopf wrote:
>
> > Dear Colleagues,
> >
> > I would like to commence the work on defining a Comfort Noise Generation
> > scheme for speech codecs without a built-in algorithm.  The proposed steps
> > to complete this work as outlined by the Rapporteur of Q.19 include :
> >
> > (1)     Complete the Performance Requirements and Objectives
> > (2)     Draft preliminary test methodology and ask Q.14/12 for comments
> > (3)     Complete the Test Methodology
> > (4)     Collect candidate algorithms for CNG
> > (5)     Performance assessment for the candidates
> > (6)     Select a candidate and define the algorithm (Draft Appendix to 
> G.711) .
> >
> > Because the work is urgently needed by WP2/16 and the IETF, the goal should
> > be to have a draft appendix ready for approval at the SG16 meeting in Feb.
> > 2000.  The plan is to use  wp3audio as the reflector for discussions.
> >
> > Please offer any comments or suggestions with respect to this proposed
> > procedure.
> > I will circulate a revised draft of the Terms of Reference for review and
> > comments shortly.
> >
> > Best regards,
> >
> > Rob.
> >
> > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> > Rob Zopf
> > Principal Engineer
> > Speech Technology Group
> > Lucent Technologies, INS
> >
> > Phone   : (732) 578-3207
> > Fax             : (732) 578-3213
> > Email           : zopf@lucent.com
> > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~




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At 06:49 AM 10/21/99 -0400, Internet-Drafts@ietf.org wrote:
>A New Internet-Draft is available from the on-line Internet-Drafts directories.
>This draft is a work item of the Audio/Video Transport Working Group of the 
>IETF.
>
>	Title		: A More Loss-Tolerant RTP Payload Format for MP3 Audio
>	Author(s)	: R. Finlayson
>	Filename	: draft-ietf-avt-rtp-mp3-00.txt
>	Pages		: 7
>	Date		: 20-Oct-99
>	
>While the RTP payload format defined in RFC 2250 is generally applicable
>to all forms of MPEG audio or video, it is less suitable for MPEG 1 or 
>2,layer III audio (commonly known as 'MP3').  The reason for this is that an 
>MP3 frame is not a true 'Application Data Unit' - it contains a
>back-pointer to data in earlier frames, and so cannot be decoded
>independently of these earlier frames.  Because RFC 2250 defines that
>packet boundaries coincide with frame boundaries, it handles packet loss
>inefficiently when carrying MP3 data.  The loss of an MP3 frame will
>render some data in previous (or future) frames useless, even if they are 
>received without loss.
>In this document we define a new RTP payload format for MP3 audio.
>The new format is essentially the same as that defined in RFC 2250, except 
>for a data-preserving rearrangement of the original MPEG frames, so that 
>packet boundaries now coincide with true MP3 'Application Data Units'. This 
>new format is therefore more data efficient in the face of packet
>loss.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-mp3-00.txt
>
>Internet-Drafts are also available by anonymous FTP. Login with the username
>"anonymous" and a password of your e-mail address. After logging in,
>type "cd internet-drafts" and then
>	"get draft-ietf-avt-rtp-mp3-00.txt".
>
>A list of Internet-Drafts directories can be found in
>http://www.ietf.org/shadow.html 
>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>
>Internet-Drafts can also be obtained by e-mail.
>
>Send a message to:
>	mailserv@ietf.org.
>In the body type:
>	"FILE /internet-drafts/draft-ietf-avt-rtp-mp3-00.txt".
>	
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From rem-conf-request@es.net  Thu Oct 21 20:54:38 1999
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From: Bill Nowicki <BNowicki@Omneon.com>
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Subject: RE: I-D ACTION:draft-ietf-avt-dv-video-01.txt
Date: Thu, 21 Oct 1999 17:46:41 -0700
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Generally the protocol is very nice and simple. However, the document has
some minor problems.

There are two spaces between "Carsten" and "Bormann" on the first page?

Be consistent about "IEC 61834" having a space between the letters and
numbers. Similarly, "IEEE 1394" is the convention, not "IEEE1394". Also put
a space before open parentheses consistently.

The designations "D-7" and "D-9" are tape formats. They correspond to
products with the DVCPRO and Digital-S trademarks respectively; is the
policy to avoid trademarks in IETF standards? If not, mentioning them might
be useful. The professional 25Mbps variants are generally the same DV
stream, just on different physical tape, so it is not clear the full matrix
is relevant to an RTP payload, but I suppose it does not hurt. There are
some subtle differences in audio support, 625 line color model, etc.

Might mention somewhere that SD-VCR stands for "Standard Definition Video
Cassette Recorder", etc. I do not know exactly what "SDL" stands for, since
nobody uses it that I know of.

Section 2 and the first paragraph of section 3 have many English errors, and
are redundant. Suggested rewording:

2. Introduction

   This document specifies payload formats for encapsulating 
   both consumer and professional use DV format data
   streams into the Real-time Transport Protocol (RTP), version 2 [6].
   DV compression audio and video formats were designed for helical-scan 
   magnetic tape media. The DV standard for consumer market devices,
   the IEC 61883 and 61834 series, includes many aspects of
   consumer use digital video, including mechanical specifications of a
   cassette, magnetic recording format, error correction on the
   magnetic tape, DCT video encoding format, and audio encoding
   format[1]. The digital interface part of IEC 61883 defines an
   interface  on an IEEE 1394 network[2,3]. This specification set supports
   several video formats: SD-VCR (Standard Definition), HD-VCR (High
   Definition), SDL-VCR (Standard Definition - Long), PALPlus, DVB (Digital
Video
   Broadcast) and ATV (Advanced Television). North American formats are
indicated
   with a number of lines and "/60", while European formats use "/50".
   DV standards extended for professional use were published by
   SMPTE as 306M and 314M, for higher color resolution and faster bit
rates[4,5].

   IEC 61834 also includes magnetic tape recording for digital TV
   broadcasting systems (such as DVB and ATV) that use MPEG2 encoding.
   The payload format for encapsulating MPEG2 into RTP has already been
   defined in RFC 2250[7] and others. 

   Consequently, the payload specified in this document will
   support six video formats of the IEC standard: SD-VCR (525/60,
   625/50), HD-VCR (1125/60, 1250/50) and SDL-VCR (525/60, 625/50), and
   six of the SMPTE standards: 306M (525/60, 625/50), 314M 25Mbps (525/60,
625/50)
   and 314M 50Mbps (525/60, 625/50). In the future it can be extended
   for other high-definition formats.

   Throughout this specification, we make extensive use of the
   terminology of IEC and SMPTE standards. The reader should consult the
   original references for definitions of these terms.

3. DV format encoding

   <delete first paragraph> The DV format the DCT compression technique only

   within each frame, contrasting with the interframe compression
   of the MPEG video standards [ref]. 

"Audio data part" should be "The audio data"

several DIF sequence. should be "several DIF sequences."

Section 4.1

"in VAUX. Because" should be "in VAUX, because"

   ... Even if VAUX data is
   received, the receiver cannot obtain the attribute until the video
   encoding format is determined.

This sentence is confusing, and I think is just trying to say what has
already been said - use payload types to determine format, not the DV
stream. It is ambiguous, however, in the bundled case, if just changing the
audio format counts as a payload type change? Looks like "no" in this draft.

Section 4.2:

It is not clear if Header and Subcode DIF blocks may, should or must be
sent?

How can the sender omit AAUX blocks in a bundled stream? If so, then all the
audio information must be specified in ugly "a-encode" lines like the
previous draft, but those are now gone? So I would say remove the reference
to AAUX here. They are not separate DIF blocks anyway, but embedded in the
Audio blocks, so just passing them through sounds like the best idea.

Section 5:

   ... SDP carries several m=?? lines which is for media type ...
should be:
   ... SDP carries several m=?? lines, one for each media type ...

Section 5.2:

"Too many audio formats" omit "Too", please be objective (thought I agree!).

flame --> frame

Reword middle of paragraph to:

   ... These attributes are carried as
   AAUX data within audio DIF blocks. Describing all these attributes
   with SDP would require large SDP descriptions to enumerate
   all combinations.  Therefore, the SDP entry to describe bundled streams
   does not include audio format parameters.

Does it include video format parameters? MAY? The example uses "encode:"
instead of "v-encode:" why be inconsistent?

Looking forward to implementations, 
	Bill Nowicki
	Omneon Video Networks
	408-558-2126



From rem-conf-request@es.net  Thu Oct 21 23:12:04 1999
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At 05:24 PM 10/21/99 -0700, you wrote:
>At 06:49 AM 10/21/99 -0400, Internet-Drafts@ietf.org wrote:
>>A New Internet-Draft is available from the on-line Internet-Drafts
directories.
>>This draft is a work item of the Audio/Video Transport Working Group of the 
>>IETF.
>>
>>	Title		: A More Loss-Tolerant RTP Payload Format for MP3 Audio
>>	Author(s)	: R. Finlayson
>>	Filename	: draft-ietf-avt-rtp-mp3-00.txt
>>	Pages		: 7
>>	Date		: 20-Oct-99
>>	
>>While the RTP payload format defined in RFC 2250 is generally applicable
>>to all forms of MPEG audio or video, it is less suitable for MPEG 1 or 
>>2,layer III audio (commonly known as 'MP3').  The reason for this is that
an 
>>MP3 frame is not a true 'Application Data Unit' - it contains a
>>back-pointer to data in earlier frames, and so cannot be decoded
>>independently of these earlier frames.  Because RFC 2250 defines that
>>packet boundaries coincide with frame boundaries, it handles packet loss
>>inefficiently when carrying MP3 data.  The loss of an MP3 frame will
>>render some data in previous (or future) frames useless, even if they are 
>>received without loss.
>>In this document we define a new RTP payload format for MP3 audio.
>>The new format is essentially the same as that defined in RFC 2250, except 
>>for a data-preserving rearrangement of the original MPEG frames, so that 
>>packet boundaries now coincide with true MP3 'Application Data Units'. This 
>>new format is therefore more data efficient in the face of packet
>>loss.
>>
>>A URL for this Internet-Draft is:
>>http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-mp3-00.txt
>>
>>Internet-Drafts are also available by anonymous FTP. Login with the username
>>"anonymous" and a password of your e-mail address. After logging in,
>>type "cd internet-drafts" and then
>>	"get draft-ietf-avt-rtp-mp3-00.txt".
>>
>>A list of Internet-Drafts directories can be found in
>>http://www.ietf.org/shadow.html 
>>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>
>>
>>Internet-Drafts can also be obtained by e-mail.
>>
>>Send a message to:
>>	mailserv@ietf.org.
>>In the body type:
>>	"FILE /internet-drafts/draft-ietf-avt-rtp-mp3-00.txt".
>>	
>>NOTE:	The mail server at ietf.org can return the document in
>>	MIME-encoded form by using the "mpack" utility.  To use this
>>	feature, insert the command "ENCODING mime" before the "FILE"
>>	command.  To decode the response(s), you will need "munpack" or
>>	a MIME-compliant mail reader.  Different MIME-compliant mail readers
>>	exhibit different behavior, especially when dealing with
>>	"multipart" MIME messages (i.e. documents which have been split
>>	up into multiple messages), so check your local documentation on
>>	how to manipulate these messages.
>>		
>>		
>>Below is the data which will enable a MIME compliant mail reader
>>implementation to automatically retrieve the ASCII version of the
>>Internet-Draft.
>>Content-Type: text/plain
>>Content-ID:	<19991020133259.I-D@ietf.org>
>>
>>ENCODING mime
>>FILE /internet-drafts/draft-ietf-avt-rtp-mp3-00.txt
>>
>><ftp://ftp.ietf.org/internet-drafts/draft-ietf-avt-rtp-mp3-00.txt>
> 



From rem-conf-request@es.net  Fri Oct 22 00:54:33 1999
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Stephen Casner wrote:
> 
> AVT working group:
> 
> I have a few questions regarding changes to the RTP spec.  I'm
> interested in any comments you may have now, and we can discuss these
> at the upcoming meeting as well.
> 
> 1. At the Oslo IETF meeting, and as reported in the minutes of that
> meeting, we agreed to change the representation of the loop/collision
> detection algorithm to pseudo-C.  This was part of the question, when
> a new source transport address is seen for a known SSRC id, whether
> packets from the old address should be kept and from the new address
> ignored, as is currently specified, or vice versa.  The motivation for
> changing the spec is to accommodate mobile hosts that may change their
> address.  Since following the new source requires hysteresis in case
> there really are two sources colliding, I propose to not add any
> additional complexity to the pseudo-code algorithm, but to add in the
> text the following statement:
> 
>     For applications in which sources may change addresses, the RTP
>     implementation SHOULD modify the collision detection algorithm to
>     accept packets from the new source transport address.  To guard
>     against flip-flopping between addresses if a genuine collision
>     does occur, the algorithm should include some means to detect this
>     case and avoid switching.

The problem is that your RTP implementation above may not be mobile. It
may be just a normal host talking to a mobile. In this case, the normal
host does have to implement this modified collision detection algorithm.
So, how would a non-mobile host know which algorithm to implement?

> 
> 2. As discussed in August, if a source has not been sending data for a
> while, so it is sending RR's with a fairly large interval according to
> the receiver fraction of the RTCP bandwidth, and then it begins to
> transmit, it may be quite a while before the RTCP interval expires and
> the first SR is sent.  The change is to add a statement to 6.3.8 that
> a transition of we_sent from false to true SHOULD trigger a "reverse
> reconsideration" recalculation of the RTCP interval.

Yes.

> 
> 3. In scenarios with one or a small number of data senders and many
> receivers, the sender's RTCP interval may be much shorter than that of
> the receivers.  Therefore, 6.3.5 on timing out an SSRC needs to be
> based on the RTCP interval using the receiver fraction rather than the
> sender fraction, even for senders.

Yes.

> 
> 4. In Oslo we decided _not_ to change the calculation of avg_rtcp_size
> to include only the RTCP packets being transmitted and exclude those
> received from other sources, even though there is a potential for
> unfair use of RTCP bandwidth, because it would complicate many parts
> of the algorithm without a clear gain.

Agree again.

> 
> A related question has occurred to me just recently:  If an RTCP
> compound packet includes information from more than one host, as in
> the case where an RTP mixer aggregates SDES information from multiple
> sources, should the size of that packet be divided by the number of
> sources before being figured into the average?

Good catch; I believe it does need to be divided in this way. Otherwise,
if the packet size is more than 5 times the average size, there will be
constant timeouts. Basically, the inverval between appearances of data
from a particular user in the mixer must be the same as if they were not
being mixed. So, if a mixer mixes 3 people, the interval between RTCP
packets containing data for user 1 should be the same as if user 1 were
not in the mixer.

Hmm; another way to divide this is to actually allocate N times the
bandwidth if N users are being mixed, but don't divide the packet size
by N. I think this is more or less equivalent to what you suggest.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf-request@es.net  Fri Oct 22 01:07:25 1999
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From: Stephen Casner <casner@cisco.com>
To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
cc: rem-conf@es.net
Subject: Re: RTP spec questions
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On Fri, 22 Oct 1999, Jonathan Rosenberg wrote:

> >     For applications in which sources may change addresses, the RTP
> >     implementation SHOULD modify the collision detection algorithm to
> >     accept packets from the new source transport address.  To guard
> >     against flip-flopping between addresses if a genuine collision
> >     does occur, the algorithm should include some means to detect this
> >     case and avoid switching.
> 
> The problem is that your RTP implementation above may not be mobile. It
> may be just a normal host talking to a mobile. In this case, the normal
> host does have to implement this modified collision detection algorithm.
> So, how would a non-mobile host know which algorithm to implement?

Right.  I was attempting to distinguish by using "application" and
"implementation" that all nodes, both mobile and non-mobile, which
participate in an application like IP telephony where addresses might
change on the fly are affected by this consideration.  Would
"scenarios" do better than "applications"?

> > A related question has occurred to me just recently:  If an RTCP
> > compound packet includes information from more than one host, as in
> > the case where an RTP mixer aggregates SDES information from multiple
> > sources, should the size of that packet be divided by the number of
> > sources before being figured into the average?
> 
> Good catch; I believe it does need to be divided in this way. Otherwise,
> if the packet size is more than 5 times the average size, there will be
> constant timeouts.

I'm not sure that's true.  Everyone should be averaging over the same
set of packets, so the intervals should be roughly consistent.  Well,
execept that the estimator gain is 1/16 so it could bounce around if
the big packet was only one out of many.  My concern was more that the
average would be wrong -- too large, which means intervals longer than
they should be, which is the safer direction.
							-- Steve




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On Thu, 21 Oct 1999, Bill Nowicki wrote:

:
:
:
> Looking forward to implementations, 

There are at least three implementations under way that I know of:

http://www.sfc.wide.ad.jp/DVTS

http://www.it.kth.se/~nv91-tob/Report/Prototype/

And Carsten Bormann has something brewing. I have no pointer to that 
work, though.

/Tobias
............................................................................
Tobias Öbrink                 Grad. Stud. Dept. of Teleinformatics, KTH
Phone numbers:				Paper mail address:
	+46 8 752 14 75 (IT, work)		KTH-ELECTRUM/204
        +46 8 751 17 93 (Fax)			S-164 40 Kista
						Sweden
E-mail: tobias@it.kth.se	URL: http://www.it.kth.se/~nv91-tob/
See if I'm logged in: http://www.it.kth.se/~nv91-tob/notice.html





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Date: Fri, 22 Oct 1999 10:11:19 +0200
To: rem-conf@es.net
From: Harald Tveit Alvestrand <Harald@Alvestrand.no>
Subject: Fwd: Liaison Statment to IETF (SC 29 N 3290)
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The attached comments to the MPEG RTP spec were contributed from our
liaison with ISO SC29. They are forwarded without much comment from me.

                       Harald


>Date: Fri, 22 Oct 1999 14:29:06 +0900
>From: Narumi Hirose <hirose@itscj.ipsj.or.jp>
>To: Harald.T.Alvestrand@uninett.no
>Cc: Barry <bgh@research.att.com>, Leonardo <leonardo.chiariglione@CSELT.IT>,
>         kogure@drl.mei.co.jp
>Subject: Liaison Statment to IETF (SC 29 N 3290)
>Reply-To: hirose@itscj.ipsj.or.jp
>X-Mailer: Becky! ver 1.25.07
>
>Dear Mr. Alvestrand,
>
>In accordance with Resolution 2.8.3 taken at the forty-ninth
>ISO/IEC JTC 1/SC 29/WG 11 meeting, 1999-10-04/08, Melbourne,
>Australia, I have pleasure in forwarding the liaison statement
>to IETF.
>
>SC 29 N 3290 [SC 29/WG 11 N 2987]:
>Liaison Statement from SC 29/WG 11 to IETF on MPEG-4 on
>the Internet
>Attachment: SC 29/WG 11 N 3021
>Study on Internet draft for the carriage of MPEG-4 on IP
>
>Should you have any questions, please do not hesitate
>to contact me.
>
>Thank you for your cooperation.
>
>Regards,
>
>Narumi
>===============================================
>Narumi Hirose
>Secretariat, ISO/IEC JTC 1/SC 29
>IPSJ/ITSCJ Room 308-3 Kikai-Shinko-Kaikan Bldg
>3-5-8 Shiba-koen Minato-ku Tokyo 105-0011 Japan
>Tel: +81-3-3431-2808  Fax: +81-3-3431-6493
>E-mail: hirose@itscj.ipsj.or.jp
>===============================================
>

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AAAAAAAAAAAgAAAAAAAAADI5bjMyOTAxLmRvY1BLAQIUABQAAAAIAIBJVicbtma2yggAAL8UAAAM
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--=====================_508417375==_
Content-Type: text/plain; charset="us-ascii"; format=flowed

--
Harald Tveit Alvestrand, Maxware, Norway
Harald.Alvestrand@maxware.no

--=====================_508417375==_--




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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: A More Loss-Tolerant RTP Payload Format for MP3 Audio
	Author(s)	: R. Finlayson
	Filename	: draft-ietf-avt-rtp-mp3-01.txt
	Pages		: 7
	Date		: 21-Oct-99
	
While the RTP payload format defined in RFC 2250 is generally applicable
to all forms of MPEG audio or video, it is less suitable for MPEG 1 or 2, layer III audio (commonly known as 'MP3').  The reason for this is that an MP3 frame is not a true ''Application Data Unit' - it contains a
back-pointer to data in earlier frames, and so cannot be decoded
independently of these earlier frames.  Because RFC 2250 defines that
packet boundaries coincide with frame boundaries, it handles packet loss
inefficiently when carrying MP3 data.  The loss of an MP3 frame will
render some data in previous (or future) frames useless, even if they are received without loss.

In this document we define a new RTP payload format for MP3 audio.
The new format is essentially the same as that defined in RFC 2250, except for a data-preserving rearrangement of the original MPEG frames, so that packet boundaries now coincide with true MP3 'Application Data Units'.
This new format is therefore more data efficient in the face of packet
loss.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-mp3-01.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
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type "cd internet-drafts" and then
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A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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Send a message to:
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From rem-conf-request@es.net  Fri Oct 22 07:30:58 1999
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Bill,

Thanks for useful comment.

From: Bill Nowicki <BNowicki@Omneon.com>
Subject: RE: I-D ACTION:draft-ietf-avt-dv-video-01.txt
Date: Thu, 21 Oct 1999 17:46:41 -0700
Message-ID: <3A54A80C464CD311BAAA00A0C9C85E6704532C@grumpy.omneon.com>

> Generally the protocol is very nice and simple. However, the document has
> some minor problems.
> 
> There are two spaces between "Carsten" and "Bormann" on the first page?
> 
> Be consistent about "IEC 61834" having a space between the letters and
> numbers. Similarly, "IEEE 1394" is the convention, not "IEEE1394". Also put
> a space before open parentheses consistently.
> 

We will correct them in the next publication.

> The designations "D-7" and "D-9" are tape formats. They correspond to
> products with the DVCPRO and Digital-S trademarks respectively; is the
> policy to avoid trademarks in IETF standards? If not, mentioning them might
> be useful. 

I agree it, the name DVCPRO and Digital-S are well known compared with
SMPTE standard number. I don't know the exact attitude of IETF for
trademark. If we can import it without trouble, it is better to clearly
specify trademark.

Does someone know the IETF attitude or the reference concerning 
the trademark issue ?

> The professional 25Mbps variants are generally the same DV
> stream, just on different physical tape, so it is not clear the full matrix
> is relevant to an RTP payload, but I suppose it does not hurt. There are
> some subtle differences in audio support, 625 line color model, etc.

I think generally the same. But system data treatment is different,
it is safe to distinguish them. As you mentioned, the picture sampling
structures in 625 are not the same, cunsumer uses 4:2:0 and
professional is 4:1:1. I think this difference is not small.
In the implementation, same CODEC hardware and software can be used
with mode switching.

> 
> Might mention somewhere that SD-VCR stands for "Standard Definition Video
> Cassette Recorder", etc. I do not know exactly what "SDL" stands for, since
> nobody uses it that I know of.
> 

The relation between SDL and SD is the same between LP and standard
on DAT. That standarization is working in IEC SC100B/PT 61834-6.
The SDL is also specified in the BlueBook.

> Section 2 and the first paragraph of section 3 have many English errors, and
> are redundant. Suggested rewording:
> 

Thanks, we will incorporate your input in the next draft.

..snip..

> 
> Section 4.1
> 
> "in VAUX. Because" should be "in VAUX, because"
> 
>    ... Even if VAUX data is
>    received, the receiver cannot obtain the attribute until the video
>    encoding format is determined.
> 
> This sentence is confusing, and I think is just trying to say what has
> already been said - use payload types to determine format, not the DV
> stream. It is ambiguous, however, in the bundled case, if just changing the
> audio format counts as a payload type change? Looks like "no" in this draft.
> 

In bundled case, only audio format changing is permitted. How about
remove sentence bellow as ?

     ... Although VAUX data contains some encoding attributes of
     the stream, the sender MUST NOT expect to notify the receiver
     of an encoding format change with the information included
     in VAUX because the packet carrying the VAUX data might be lost.

And changes the sentence concerning VAUX:

     ...Even if the VAUX data is received, the receiver cannot obtain
     the attributes until the video encoding format is determined
     because the VAUX data only represents sub-attribute information
     which relies on the video encoding format.

> Section 4.2:
> 
> It is not clear if Header and Subcode DIF blocks may, should or must be
> sent?
> 

Add sentence as:

    ...Other system data header and subcode DIF block MAY also be
    sent. When header and subcode DIF blocks is sent, the both blocks
    MUST be included in the video stream.

> How can the sender omit AAUX blocks in a bundled stream? If so, then all the
> audio information must be specified in ugly "a-encode" lines like the
> previous draft, but those are now gone? So I would say remove the reference
> to AAUX here. They are not separate DIF blocks anyway, but embedded in the
> Audio blocks, so just passing them through sounds like the best idea.
> 

I think our draft only say as "... send null AAUX information 
and omit VAUX DIF block....". If this sentence leads other people in
confusion, we have to change it.

> Section 5:

..snip..

> 
> Reword middle of paragraph to:
> 
>    ... These attributes are carried as
>    AAUX data within audio DIF blocks. Describing all these attributes
>    with SDP would require large SDP descriptions to enumerate
>    all combinations.  Therefore, the SDP entry to describe bundled streams
>    does not include audio format parameters.
> 

We will change to:

    This includes audio attribute information carried as AAUX data within
    audio DIF blocks. Many audio format attributes are defined in DV, such
    as sampling, quantization, number of audio channels, channel
    assignment, language, emphasis and the picture frame locking.
    Describing all these attributes with SDP would require large SDP
    descriptions to enumerate all combinations.  Therefore, the SDP entry
    to describe bundled streams does not include audio format parameters.

> 
> Looking forward to implementations, 

If we get stable implementation, we will demonstrate in opportunity.

                                              ikob@koganei.wide.ad.jp



From rem-conf-request@es.net  Fri Oct 22 10:17:35 1999
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From: Alexander Tulai - icpd <alexander_tulai@Mitel.COM>
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Hi,

We put together an internet draft that talks about the Nx64 problem
(transporting
multiple 64 kb/s streams between two gateways).
This is a subset of the multiplexing problem and, because all
multiplexed streams are
identically coded, further bandwidth savings could be achieved with a
different
RTP payload type. The implementation of this method is very simple,
robust, and
allows for an easy mapping to existing dynamic Nx64 methods used in ATM
which
could be very useful in equipments bridging IP and ATM networks.
In addition, the method proposed allows for silence supression and the
continuous
sending of the CN packet payload (see draft-ietf-avt-profile-new-06.txt)
that allows
the receiver to better follow the noise level at the source.

The draft is an individual submission, and will appear in the archives
shortly
with the title: draft-ietf-avt-dynamic-nx64-00.txt
Until then, to pick up a copy ftp to "semicon.mitel.com" and cd to
"incoming"
directory.

Comments welcome.

Thanks,
Alexander Tulai
Mitel Corporation
Ph: (613) 592-2122
Fax: (613) 592-4784
Email: alexander_tulai@mitel.com






From rem-conf-request@es.net  Fri Oct 22 12:12:10 1999
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From: "Andrea Basso" <basso@research.att.com>
To: "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>, <rem-conf@es.net>
Subject: Submission of updated draft : draft-ietf-avt-rtp-mpeg4-02.txt
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The updated draft on the MPEG-4 RTP payload format should be appearing
shortly in the on-line IETF I-D archives. It
can also be found at
http://www.research.att.com/~mrc/draft-ietf-avt-rtp-mpeg4-02.txt


Any comment is more than welcome.

-A

Andrea Basso, AT&T Labs Research
Room 3-219
100 Shultz Drive, Red Bank,NJ 07701
ph:  (732) 345-3302
fax  (732) 345-3033
basso@research.att.com





From rem-conf-request@es.net  Fri Oct 22 14:33:35 1999
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To: Florissa Colina <florissa@gumby.CS.Berkeley.EDU>
From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 10/27 Supporting Multicast Services in the Emerging Internet2
  Network Infrastructure The View from Minus Six Feet  -- Ken Lindahl,
  Communication and Network Services U.C. Berkeley 
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Berkeley Multimedia, Graphics and Interfaces Seminar

Supporting Multicast Services in the Emerging Internet2 Network
Infrastructure The View from Minus Six Feet

Wednesday October 27, 1999, 
1:00-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 


Ken Lindahl 
Communication and Network Services U.C. Berkeley  

This seminar will cover the architecture of networking @ Berkeley and
support for multicast @ Berkeley and Internet2.

---------------

The seminar is broadcast live on the Internet Mbone and as a Real
Networks G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player:

rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
   http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving
the announcement you may be able to receive the session by manually
configuring the client programs ('vic', and 'vat') with the session
addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to
replays of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf-request@es.net  Fri Oct 22 14:39:13 1999
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Date: Fri, 22 Oct 1999 20:39:19 +0200
To: Zoran Kostic <kostic@research.att.com>, pilc@grc.nasa.gov, rem-conf@es.net
From: Mikael Degermark <micke@cdt.luth.se>
Subject: Re: Header Stripping in Cellular
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Zoran, 

At the coming IETF in Washington DC, there will be a BOF discussing
robust header compression for cellular. One of the proposals to be 
presented at that BOF can compress RTP/UDP/IP down to 2 bytes (or
possibly less in coming releases!) even in the presense of substantial
loss on the link. 

Please come to the BOF. Even if we rehash this here, we will have to come
to some agreement as to what the IETF should do about header stripping 
and similar ideas. The BOF is a good venue to hash this out!

Cheers, 

Mikael Degermark (who will chair the BOF)

(I've commented your letter at some places below.)

At 12:37 PM 99/10/22 -0400, Zoran Kostic wrote:
>Header Stripping in Cellular
>
>This topic was touched on briefly before, and the consensus was
>Do not allow it to be put into any standard !!
>
>Let me be the devil's advocate, which does not mean that the principal
>idea appeals to me any more than to anyone else on this list.
>
>I am afraid that  header stripping will almost inevitably be included
>at least as an option in standards that deal with cellular. And service
>providers will probably choose that option as a default operation mode.
>
>This is why:
>
>* Cellular service providers have paid a LOT for bandwidth
>* Improvements over current 2nd generation cellular systems consist
>   to a large degree of improving spectral efficiency of cellular
>   networks (bits/second/hertz/cell) - improving the capacity -
>   improving the revenue flow

With suitable header compression it is possible to get within less than a
dB from circuit-switched voice. So the incentives aren't quite so large
as you say. 

>* Pretty much every single bit is accounted for in framing of RLPs,
>   speech blocks etc...
>* If there is even a remote chance that some bit does not have to be
>carried
>   over the air, it should not be carried over the air, because spectral
>
>   efficienty suffers, cellular capacity gets hurt, revenue and profit
>go down
>
>The questions we have to ask us are:
>
>1) What might be the highest compression rate achievable for TCP/IP
>headers ?
>    Even it is 10, I believe that stripping will be what service
>providers
>    would opt for. And can we even come close to 10 ?

Do you mean TCP/IP or RTP/UDP/IP?
For TCP/IP I find it difficult to go below 4 bytes. 
Which is 10.

For RTP/UDP/IP 2 bytes or less seems within reach (provided the UDP
checksumming issues can be resolved in a reasonable manner.)
Which would be 20. 

>2) If we buy into the argument that header stripping will be legitimized
>
>    regardless of our opinion, what should IETF do about it ? Maybe
>there
>    should be something that from spectral efficiency mimicks header
>    stripping but is more acceptable to this community (again I am not
>    sure that compression will do the job).

Please come to the BOF!

>3) Heretic thought: Cellular services are a kinda special, since the
>base does many things
>    for its mobiles anyway... do we have to be exclusive on the topic of
>header
>    stripping and recreation ? Maybe we should come up
>    with recommnedations  on how should header stripping and recreation
>    be done ??

Perhaps if one could somehow view the cellular link as a kind of RTP mixer
it could be legal to fiddle with the IP/UDP headers. The RTP header must,
as far as I can see, be recreated *exactly*. 

One thing the cellular people seem to like is "All-IP" or "IP all-the-way"
which 
implies that the data is carried using IP end-to-end. If header stripping is 
used, that seems to be a mis-nomer. (Not that that would stop any marketing
people from using it...)

Personally, I believe it is possible to design a sufficiently good header
compression 
scheme that will satisfy both die-hard IP people and the spectrum-efficiency
hungry cellular people! 

Micke






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 <<draft-ietf-avt-rtp-mib-07.txt>> 

This draft contains typo fixes of errors found in WG Last call.

Bill
______________________________________________
Bill Strahm        Programming today is a race between
bill.strahm@      software engineers striving to build
intel.com           bigger and better idiot-proof programs,
(503) 264-4632   and the Universe trying to produce
            bigger and better idiots.  So far, the
                        Universe is winning.--Rich Cook
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Audio Video Transport Group                          Mark Baugher
Internet-Draft                                        Intel Corp.
Expires April 27, 2000                             Irina Suconick
                                                VideoServer Corp.
                                                      Bill Strahm
                                                      Intel Corp.
                                                 October 27, 1999

                 Real-Time Transport Protocol
                 Management Information Base
               <draft-ietf-avt-rtp-mib-07.txt>                    =20


Status of this Memo

This document is an Internet-Draft and is in full conformance
with all provisions of section 10 of RFC2026.  Internet-Drafts are
working documents of the Internet Engineering Task Force
(IETF), its areas, and its working groups.  Note that other
groups may also distribute working documents as Internet-Drafts.
Internet-Drafts are draft documents valid for a maximum of six
months.  Internet-Drafts may be updated, replaced, or made
obsolete by other documents at any time.  It is not
appropriate to use Internet-Drafts as reference material or to
cite them other than as a ``working draft'' or ``work in progress.''
To learn the current status of any Internet-Draft, please
check the 1id-abstracts.txt listing contained in the Internet-
Drafts Shadow Directories on ftp.ietf.org, nic.nordu.net,
venera.isi.edu, or munnari.oz.au.

Copyright Notice
   Copyright (C) The Internet Society (1999).  All Rights Reserved.


Abstract

This memo defines a portion of the Management Information Base
(MIB) for use with network management protocols in the Internet
community.  In particular, it defines objects for managing Real-Time=20
Transport Protocol(RTP) systems [RFC1889].
















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Table of Contents


1 The Network Management Framework ................................   3
2 Overview ........................................................   4
2.1 Components ....................................................   4
2.2 Applicability of the MIB to RTP System Implementations ........   5
2.3 The Structure of the RTP MIB ..................................   5
3 Definitions .....................................................   7
4 Security Issues .................................................  33
5 Acknowledgements ................................................  34
6 Intellectual Property ...........................................  34
7 References ......................................................  34
8 Authors Addresses ...............................................  37
9 Full Copyright Statement ........................................  37
																  =20

































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1.  The SNMP Management Framework

The SNMP Management Framework presently consists of five major
components:

    o   An overall architecture, described in RFC 2571 [RFC2571].
   =20
    o   Mechanisms for describing and naming objects and events for the
        purpose of management. The first version of this Structure of
        Management Information (SMI) is called SMIv1 and described in
        STD 16, RFC 1155 [RFC1155], STD 16, RFC 1212 [RFC1212] and RFC
        1215 [RFC1215]. The second version, called SMIv2, is described
        in STD 58, RFC 2578 [RFC2578], RFC 2579 [RFC2579] and RFC 2580
        [RFC2580].
   =20
    o   Message protocols for transferring management information. The
        first version of the SNMP message protocol is called SNMPv1 and
        described in STD 15, RFC 1157 [RFC1157]. A second version of =
the
        SNMP message protocol, which is not an Internet standards track
        protocol, is called SNMPv2c and described in RFC 1901 [RFC1901]
        and RFC 1906 [RFC1906]. The third version of the message
        protocol is called SNMPv3 and described in RFC 1906 [RFC1906],
        RFC 2572 [RFC2572] and RFC 2574 [RFC2574].
   =20
    o   Protocol operations for accessing management information. The
        first set of protocol operations and associated PDU formats is
        described in STD 15, RFC 1157 [RFC1157]. A second set of
        protocol operations and associated PDU formats is described in
        RFC 1905 [RFC1905].
   =20
    o   A set of fundamental applications described in RFC 2573
        [RFC2573] and the view-based access control mechanism described
        in RFC 2575 [RFC2575].

A more detailed introduction to the current SNMP Management Framework
can be found in RFC 2570 [RFC2570].

Managed objects are accessed via a virtual information store, termed
the Management Information Base or MIB.  Objects in the MIB are
defined using the mechanisms defined in the SMI.










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This memo specifies a MIB module that is compliant to the SMIv2. A
MIB conforming to the SMIv1 can be produced through the appropriate
translations. The resulting translated MIB must be semantically
equivalent, except where objects or events are omitted because no
translation is possible (use of Counter64). Some machine readable
information in SMIv2 will be converted into textual descriptions in
SMIv1 during the translation process. However, this loss of machine
readable information is not considered to change the semantics of the   =

MIB.

2. Overview

An "RTP System" may be a host end-system that runs an application
program that sends or receives RTP data packets, or it may be an
intermediate-system that forwards RTP packets.  RTP Control=20
Protocol (RTCP) packets are sent by senders and receivers to=20
convey information about RTP packet transmission and reception =
[RFC1889].
RTP monitors may collect RTCP information on senders and receivers
to and from an RTP host or intermediate-system.

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
RFC 2119.

2.1 Components

The RTP MIB is structured around "Session," "Receiver" and "Sender"=20
conceptual abstractions.

2.1.1  An "RTP Session" is the "...association of participants
communicating with RTP.  For each participant, the session is
defined by a particular pair of destination transport addresses
(one network address plus a port pair for RTP and RTCP).  The
destination transport addresses may be common for all participants,
as in the case of IP multicast, or may be different for each, as in
the case of individual unicast addresses plus a common port pair,"
as defined in section 3 of [RFC1889].

2.1.2 A "Sender" is identified within an RTP session by a 32-bit=20
numeric "Synchronization Source," or "SSRC", value and is "...the=20
source of a stream of RTP packets" as defined in section 3 of =
[RFC1889]. =20
The sender is also a source of RTCP Sender Report packets as specified=20
in section 6 of [RFC1889].

2.1.3 A "Receiver" of a "stream of RTP packets" can be a unicast or
multicast Receiver as described in 2.1.1, above. An RTP Receiver has=20
an SSRC value that is unique to the session.  An RTP Receiver is a=20
source of RTCP Receiver Reports as specified in section 6 of [RFC1889].

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2.2 Applicability of the MIB to RTP System Implementations=20

The RTP MIB may be used in two types of RTP implementations, RTP Host
Systems (end systems)  and RTP Monitors, see section 3 of [RFC1889]. =20
Use of the RTP MIB for RTP Translators and Mixers, as defined in=20
section 7 of [RFC1889], is for further study.

2.2.1 RTP host Systems are end-systems that may use the RTP MIB=20
to collect RTP session and stream data that the host is sending or
receiving; these data may be used by a network manager to detect and
diagnose faults that occur over the lifetime of an RTP session as=20
in a "help-desk" scenario.

2.2.2 RTP Monitors of multicast RTP sessions may be third-party or
may be located in an RTP intermediate-system or in the RTP host.  RTP=20
Monitors may use the RTP MIB to collect RTP session and stream=20
statistical data; these data may be used by a network manager for
capacity planning and other network-management purposes.  An RTP=20
Monitor may use the RTP MIB to collect data to permit a network=20
manager to detect and diagnose faults in RTP sessions or to permit=20
a network manger to configure its operation.        =20

2.2.3 Many host systems will want to keep track of streams beyond what
they are sending and receiving.  In a host monitor system, a host agent =
would
use RTP data from the host to maintain data about streams it is sending
and receiving, and RTCP data to collect data about other hosts in the=20
session.  For example an agent for an RTP host that is sending a stream =
would
use data from its RTP system to maintain the rtpSenderTable, but it may
want to maintain a rtpRcvrTable for endpoints that are receiving its =
stream. =20
To do this the RTP agent will collect RTCP data from the receivers of =
its=20
stream to build the rtpRcvrTable.  A host monitor system MUST set the=20
rtpSessionMonitor object to 'true(1)', but it does not have to accept=20
management operations that create and destroy rows in its =
rtpSessionTable.

2.3  The Structure of the RTP MIB

There are six tables in the RTP MIB.  The rtpSessionTable contains
objects that describe active sessions at the host, intermediate system,
or monitor.  The rtpSenderTable contains information about senders
to the RTP session.  The rtpRcvrTable contains information about
receivers of RTP session data. The rtpSessionInverseTable,=20
rtpSenderInverseTable, and rtpRcvrInverseTable contain information
to efficiently find indexes into the rtpSessionTable, rtpSenderTable,
and rtpRcvrTable, respectively.





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The reverse lookup tables (rtpSessionInverseTable,
rtpSenderInverseTable, and rtpRcvrInverseTable) are optional tables
to help management applications efficiently access conceptual rows in=20
other tables.  Implementors of this MIB SHOULD implement these tables=20
for multicast RTP sessions when table indexes (rtpSessionIndex of=20
rtpSessionTable, rtpSenderSSRC of rtpSenderTable, and the SSRC pair in=20
the rtpRcvrTable) are not available from other MIBs.  Otherwise, the=20
management application may be forced to perform expensive tree walks=20
through large numbers of sessions, senders, or receivers.

For any particular RTP session, the rtpSessionMonitor object indicates=20
whether remote senders or receivers to the RTP session are to be =
monitored.
RTP sessions are monitored by the RTP agent that updates rtpSenderTable =

and rtpRcvrTable objects with information from RTCP reports from remote =

senders or remote receivers respectively. =20

rtpSessionNewIndex is a global object that permits a network-management
application to obtain a unique index for conceptual row creation
in the rtpSessionTable.  In this way the SNMP Set operation may
be used to configure a monitor.





























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3. Definitions
RTP-MIB DEFINITIONS ::=3D BEGIN
IMPORTS
       Counter32, Counter64, Gauge32, mib-2, Integer32,
       MODULE-IDENTITY, OBJECT-IDENTITY,=20
       OBJECT-TYPE, Unsigned32                     FROM SNMPv2-SMI
       RowStatus, TAddress,=20
       TDomain, TestAndIncr,=20
       TimeStamp, TruthValue                       FROM SNMPv2-TC
       OBJECT-GROUP, MODULE-COMPLIANCE             FROM SNMPv2-CONF
       Utf8String                                  FROM SYSAPPL-MIB
       InterfaceIndex                        	   FROM IF-MIB;

rtpMIB MODULE-IDENTITY      =20
    LAST-UPDATED "9910200000Z"
    ORGANIZATION "IETF AVT Working Group"      =20
    CONTACT-INFO
            "Mark Baugher      =20
    Postal: Intel Corporation
            2111 NE 25th Avenue             =20
            Hillsboro, OR   97124
            United States      =20
    Tel:    +1 503 466 8406
    Email:  mbaugher@intel.com              =20
	=09
            Bill Strahm
    Postal: Intel Corporation             =20
            2111 NE 25th Avenue
            Hillsboro, OR   97124             =20
            United States
    Tel:    +1 503 264 4632      =20
    Email:  bill.strahm@intel.com
             =20
            Irina Suconick      =20
    Postal: Videoserver Corporation
            63 Third Street             =20
            Burlington, MA  01803
            United States      =20
    Tel:    +1 781-505-2155
    Email:  isuconick@videoserver.com"









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        DESCRIPTION      =20
        "The managed objects of RTP systems.  The MIB is=20
        structured around three types of information.
        1. General information about RTP sessions such
           as the session address.
        2. Information about RTP streams being sent to
           an RTP session by a particular sender. =20
        3. Information about RTP streams received on an
           RTP session by a particular receiver from a=20
           particular sender.
         There are two types of RTP Systems, RTP hosts and
         RTP monitors.  As described below, certain objects
         are unique to a particular type of RTP System.   An
         RTP host may also function as an RTP monitor.
         Refer to RFC 1889, 'RTP: A Transport Protocol for=20
         Real-Time Applications,' section 3.0, for definitions."
   REVISION     "9910200000Z"
   DESCRIPTION  "Initial version of this MIB.
                 Published as RFC xxx."      -- RFC-Editor assigns xxx

::=3D { mib-2 xxx }  -- to be assigned by IANA

--
-- OBJECTS
--
rtpMIBObjects OBJECT IDENTIFIER ::=3D { rtpMIB 1 }
rtpConformance OBJECT IDENTIFIER ::=3D { rtpMIB 2 }
rtpDomains OBJECT IDENTIFIER ::=3D { rtpMIB 3 }

rtpUDPDomain OBJECT-IDENTITY
    STATUS          current      =20
    DESCRIPTION
          "RTP over UDP transport domain over IPv4.  This definition=20
          uses the transport address type, snmpUDPAddress, which is=20
          defined in SNMPv2-TM, 'Transport Mappings for Version 2 of=20
          the Simple Network Management Protocol (SNMPv2)'."
    REFERENCE       "RFC 1906, sec. 2 "       ::=3D { rtpDomains 1 }












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--
-- SESSION NEW INDEX
--
rtpSessionNewIndex OBJECT-TYPE
    SYNTAX          TestAndIncr      =20
    MAX-ACCESS      read-write
    STATUS          current      =20
    DESCRIPTION
      "This  object  is  used  to  assign  values  to rtpSessionIndex
       as described in 'Textual Conventions  for  SNMPv2'.  For an RTP=20
       system that supports the creation of rows, the  network manager=20
       would read the  object,  and  then write the value back in
       the Set that creates a new instance  of rtpSessionEntry.   If =20
       the  Set  fails with the code 'inconsistentValue,' then the=20
       process must be repeated; If the Set succeeds, then the object
       is incremented, and the  new  instance  is created according to=20
       the manager's directions.  However, if the RTP agent is not =
acting=20
       as a monitor, only the RTP agent may create conceptual rows in =
the
       RTP session table."      =20
    ::=3D { rtpMIBObjects 1 }

--
-- SESSION INVERSE TABLE
--
rtpSessionInverseTable OBJECT-TYPE      =20
    SYNTAX          SEQUENCE OF RtpSessionInverseEntry
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION
      "Maps rptSessionDomain, rtpSessionRemAddr, and
       rtpSessionLocAddr to an unsigned integer index,
       rtpSessionInverseIndex, which has the rtpSessionIndex=20
       value that maps to the same rtpSessionDomain,=20
       rtpSessionRemAddr, and rtpSessionLocAddr in the=20
       rtpSessionTable.  rtpSessionIndex can thereby be
       retrieved based upon the domain and TAdress pair
       that uniquely define an RTP Session."
    ::=3D { rtpMIBObjects 2 }











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rtpSessionInverseEntry OBJECT-TYPE      =20
    SYNTAX          RtpSessionInverseEntry
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION         =20
      "Each entry corresponds to exactly one entry in the
       rtpSessionTable - the entry containing the triple,
       rtpSessionDomain, rtpSessionRemAddr, and rtpSessionLocAddr."
    INDEX { rtpSessionDomain, rtpSessionRemAddr, rtpSessionLocAddr,=20
            rtpSessionIndex }
    ::=3D { rtpSessionInverseTable 1 }

RtpSessionInverseEntry ::=3D SEQUENCE { =20
        rtpSessionInverseStartTime     TimeStamp     =20
        }

rtpSessionInverseStartTime OBJECT-TYPE      =20
    SYNTAX          TimeStamp
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "The value of SysUpTime at the time that this row was=20
       created."      =20
    ::=3D { rtpSessionInverseEntry 1 }

-- =20
--	SESSION TABLE
--
rtpSessionTable OBJECT-TYPE
    SYNTAX          SEQUENCE OF RtpSessionEntry
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION
          "There's one entry in rtpSessionTable for each RTP session
          on which packets are being sent, received, and/or=20
          monitored."      =20
    ::=3D { rtpMIBObjects 3 }













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rtpSessionEntry OBJECT-TYPE      =20
    SYNTAX          RtpSessionEntry
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION         =20
      "Data in rtpSessionTable uniquely identify an RTP session.  A =
host=20
       RTP agent will create a read-only row for each session to which=20
       packets are being sent or received.  Rows are created by the=20
       RTP Agent at the start of a session when one or more senders or=20
       receivers are observed.  Rows created by an RTP agent are =
deleted=20
       when the session is over and there are no rtpRcvrEntry and no=20
       rtpSenderEntry for this session.  An RTP session can be =
monitored=20
       to create management information on all RTP streams being sent =
or=20
       received when the rtpSessionMonitor has the TruthValue of =
'true(1)'. =20
       An RTP monitor may permit row creation with the side effect of=20
       causing the RTP System to join the multicast session for the=20
       purposes of gathering management information  (additional =
conceptual=20
       rows are created in the rtpRcvrTable and rtpSenderTable).  Thus, =

       rtpSessionTable rows can be created for RTP session monitoring=20
       purposes.  Rows created by a management application may be =
deleted=20
       via SNMP operations by management applications.  Rows created by =

       management operations are deleted by management operations by =
setting=20
       rtpSessionRowStatus to 'destroy(6)'."
    INDEX { rtpSessionIndex }      =20
    ::=3D { rtpSessionTable 1 }

RtpSessionEntry ::=3D SEQUENCE { =20
        rtpSessionIndex         Integer32,
        rtpSessionDomain        TDomain,      =20
        rtpSessionRemAddr       TAddress,
        rtpSessionLocAddr       TAddress,
        rtpSessionIfIndex       InterfaceIndex,
        rtpSessionSenderJoins   Counter32,
        rtpSessionReceiverJoins Counter32,
        rtpSessionByes          Counter32,
        rtpSessionStartTime     TimeStamp,
       	rtpSessionMonitor       TruthValue,
       	rtpSessionRowStatus     RowStatus      =20
        }










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rtpSessionIndex OBJECT-TYPE      =20
    SYNTAX          Integer32 (1..2147483647)
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION
      "The index of the conceptual row which is for SNMP purposes
       only and has no relation to any protocol value.  There is
       no requirement that these rows are created or maintained
       sequentially."
    ::=3D { rtpSessionEntry 1 }=20

rtpSessionDomain OBJECT-TYPE
    SYNTAX          TDomain      =20
    MAX-ACCESS      read-create
    STATUS          current      =20
    DESCRIPTION
      "The transport-layer protocol used for sending or receiving
       the stream of RTP data packets on this session.
       Cannot be changed if rtpSessionRowStatus is 'active'."
    ::=3D { rtpSessionEntry 2 }

rtpSessionRemAddr OBJECT-TYPE      =20
    SYNTAX          TAddress
    MAX-ACCESS      read-create      =20
    STATUS          current
    DESCRIPTION
      "The address to which RTP packets are sent by the RTP system.  In =
an
       IP multicast RTP session, this is the single address used by all =

       senders and receivers of RTP session data.  In a unicast RTP =
session=20
       this is the unicast address of the remote RTP system. 'The=20
       destination address pair may be common for all participants, as =
in=20
       the case of IP multicast, or may be different for each, as in =
the=20
       case of individual unicast network address pairs.'  See RFC =
1889,=20
       'RTP: A Transport Protocol for Real-Time Applications,' sec. 3.  =

       The transport service is identified by rtpSessionDomain. For=20
       rtpUDPDomain, this is an IP address and even-numbered UDP Port=20
       with the RTCP being sent on the next higher odd-numbered port,=20
       see RFC 1889, sec. 5. Cannot be changed if  rtpSessionRowStatus=20
       is 'active'."    =20
    ::=3D { rtpSessionEntry 3 }









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rtpSessionLocAddr OBJECT-TYPE      =20
    SYNTAX          TAddress
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "The local address used by the RTP system.  In an IP multicast =
RTP=20
       session, rtpSessionRemAddr will be the same IP multicast address =
as=20
       rtpSessionLocAddr.  In a unicast RTP session, rtpSessionRemAddr =
and=20
       rtpSessionLocAddr will have different unicast addresses.  See =
RFC=20
       1889, 'RTP: A Transport Protocol for Real-Time Applications,' =
sec.=20
       3.  The transport service is identified by rtpSessionDomain.  =
For=20
       rtpUDPDomain, this is an IP address and even-numbered UDP Port=20
       with the RTCP being sent on the next higher odd-numbered port,=20
       see RFC 1889, sec. 5."
    ::=3D { rtpSessionEntry 4 }

rtpSessionIfIndex OBJECT-TYPE
    SYNTAX          InterfaceIndex      =20
    MAX-ACCESS      read-create
    STATUS          current      =20
    DESCRIPTION
     "The ifIndex value is set to the corresponding value=20
      from IF-MIB (See RFC 2233, 'The Interfaces Group MIB using=20
      SMIv2').  This is the interface that the RTP stream is being sent
      to or received from, or in the case of an RTP Monitor the=20
      interface that RTCP packets will be received on.  Cannot be=20
      changed if rtpSessionRowStatus is 'active'."          =20
    ::=3D { rtpSessionEntry 5 }





















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rtpSessionSenderJoins OBJECT-TYPE
    SYNTAX          Counter32      =20
    MAX-ACCESS      read-only
    STATUS          current      =20
    DESCRIPTION
      "The number of senders that have been observed to have
       joined the session since this conceptual row was created
       (rtpSessionStartTime).  A sender 'joins' an RTP
       session by sending to it.  Senders that leave and then=20
       re-join following an RTCP BYE (see RFC 1889, 'RTP: A=20
       Transport Protocol for Real-Time Applications,' sec. 6.6)
       or session timeout may be counted twice.  Every time a new
       RTP sender is detected either using RTP or RTCP, this counter
       is incremented."      =20
    ::=3D { rtpSessionEntry 6 }

rtpSessionReceiverJoins OBJECT-TYPE      =20
    SYNTAX          Counter32
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "The number of receivers that have been been observed to
       have joined this session since this conceptual row was
       created (rtpSessionStartTime).  A receiver 'joins' an RTP
       session by sending RTCP Receiver Reports to the session.
       Receivers that leave and then re-join following an RTCP BYE=20
       (see RFC 1889, 'RTP: A Transport Protocol for Real-Time=20
       Applications,' sec. 6.6) or session timeout may be counted=20
       twice."       =20
    ::=3D { rtpSessionEntry 7 }



















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rtpSessionByes OBJECT-TYPE      =20
    SYNTAX          Counter32
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "A count of RTCP BYE (see RFC 1889, 'RTP: A Transport=20
       Protocol for Real-Time Applications,' sec. 6.6) messages=20
       received by this entity."      =20
    ::=3D { rtpSessionEntry 8 }

rtpSessionStartTime OBJECT-TYPE      =20
    SYNTAX          TimeStamp
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "The value of SysUpTime at the time that this row was=20
       created."      =20
    ::=3D { rtpSessionEntry 9 }

rtpSessionMonitor OBJECT-TYPE      =20
    SYNTAX          TruthValue
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "Boolean, Set to 'true(1)' if senders or receivers in=20
       addition to the local RTP System are to be monitored using RTCP. =
=20
       RTP Monitors MUST initialize to 'true(1)' and RTP Hosts SHOULD=20
       initialize this 'false(2)'.  Note that because 'host monitor'=20
       systems are receiving RTCP from their remote participants they
       MUST set this value to 'true(1)'."
    ::=3D { rtpSessionEntry 10 }

rtpSessionRowStatus OBJECT-TYPE
    SYNTAX          RowStatus      =20
    MAX-ACCESS      read-create
    STATUS          current      =20
    DESCRIPTION
      "Value of 'active' when RTP or RTCP messages are being=20
       sent or received by an RTP System.  A newly-created
       conceptual row must have the all read-create objects
       initialized before becoming 'active'.
       A conceptual row that is in the 'notReady' or 'notInService'
       state MAY be removed after 5  minutes."
    ::=3D { rtpSessionEntry 11 }



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--
-- SENDER INVERSE TABLE
--
rtpSenderInverseTable OBJECT-TYPE      =20
    SYNTAX          SEQUENCE OF RtpSenderInverseEntry
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION
      "Maps rtpSenderAddr, rtpSessionIndex, to the rtpSenderSSRC=20
       index of the rtpSenderTable.  This table allows management=20
       applications to find entries sorted by rtpSenderAddr rather than
       sorted by rtpSessionIndex.  Given the rtpSessionDomain and=20
       rtpSenderAddr, a set of rtpSessionIndex and rtpSenderSSRC values
       can be returned from a tree walk.  When rtpSessionIndex is
       specified in the SNMP Get-Next operations, one or more=20
       rtpSenderSSRC values may be returned."
    ::=3D { rtpMIBObjects 4 }

rtpSenderInverseEntry OBJECT-TYPE      =20
    SYNTAX          RtpSenderInverseEntry
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION         =20
      "Each entry corresponds to exactly one entry in the
       rtpSenderTable - the entry containing the index pair,
       rtpSessionIndex, rtpSenderSSRC."
    INDEX { rtpSessionDomain, rtpSenderAddr, rtpSessionIndex,=20
            rtpSenderSSRC }
    ::=3D { rtpSenderInverseTable 1 }

RtpSenderInverseEntry ::=3D SEQUENCE { =20
        rtpSenderInverseStartTime     TimeStamp     =20
        }

rtpSenderInverseStartTime OBJECT-TYPE      =20
    SYNTAX          TimeStamp
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "The value of SysUpTime at the time that this row was=20
       created."      =20
    ::=3D { rtpSenderInverseEntry 1 }








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--
--  SENDERS TABLE
--
rtpSenderTable OBJECT-TYPE
    SYNTAX          SEQUENCE OF RtpSenderEntry
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION
      "Table of information about a sender or senders to an RTP=20
       Session. RTP sending hosts MUST have an entry in this table=20
       for each stream being sent.  RTP receiving hosts MAY have an=20
       entry in this table for each sending stream being received=20
       by this host.  RTP monitors create an entry for each observed=20
       sender to a multicast RTP Session as a side-effect when a=20
       conceptual row in the rtpSessionTable is made 'active' by a=20
       manager."      =20
    ::=3D { rtpMIBObjects 5 }

rtpSenderEntry OBJECT-TYPE      =20
    SYNTAX          RtpSenderEntry
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION
      "Each entry contains information from a single RTP Sender=20
       Synchronization Source (SSRC, see RFC 1889 'RTP: A Transport=20
       Protocol for Real-Time Applications' sec.6).  The session is
       identified to the the SNMP entity by rtpSessionIndex.
       Rows are removed by the RTP agent when a BYE is received
       from the sender or when the sender times out (see RFC
       1889, Sec. 6.2.1) or when the rtpSessionEntry is deleted."
    INDEX { rtpSessionIndex, rtpSenderSSRC }      =20
    ::=3D { rtpSenderTable 1 }

RtpSenderEntry ::=3D SEQUENCE {      =20
        rtpSenderSSRC           Unsigned32,
        rtpSenderCNAME          Utf8String,
        rtpSenderAddr           TAddress,
        rtpSenderPackets        Counter64,
       	rtpSenderOctets         Counter64,
       	rtpSenderTool           Utf8String,
       	rtpSenderSRs            Counter32,
       	rtpSenderSRTime         TimeStamp,      =20
       	rtpSenderPT             INTEGER,
       	rtpSenderStartTime      TimeStamp      =20
        }




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rtpSenderSSRC OBJECT-TYPE      =20
    SYNTAX          Unsigned32
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION
      "The RTP SSRC, or synchronization source identifier of the
       sender.  The RTP session address plus an SSRC uniquely=20
       identify a sender to an RTP session (see RFC 1889, 'RTP: A=20
       Transport Protocol for Real-Time Applications' sec.3)."      =20
    ::=3D { rtpSenderEntry 1 }

rtpSenderCNAME OBJECT-TYPE      =20
    SYNTAX          Utf8String
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "The RTP canonical name of the sender."      =20
    ::=3D { rtpSenderEntry 2 }

rtpSenderAddr OBJECT-TYPE      =20
    SYNTAX          TAddress
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "The unicast transport source address of the sender.  In the case =
of=20
       an RTP Monitor this address is the address that the sender is =
using=20
       to send its RTCP Sender Reports."
    ::=3D { rtpSenderEntry 3 }

rtpSenderPackets OBJECT-TYPE
    SYNTAX          Counter64      =20
    MAX-ACCESS      read-only
    STATUS          current      =20
    DESCRIPTION
      "Count of RTP packets sent by this sender, or observed by
       an RTP monitor, since rtpSenderStartTime."
    ::=3D { rtpSenderEntry 4 }











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rtpSenderOctets OBJECT-TYPE      =20
    SYNTAX          Counter64
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "Count of RTP octets sent by this sender, or observed by
       an RTP monitor, since rtpSenderStartTime."
    ::=3D { rtpSenderEntry 5 }

rtpSenderTool OBJECT-TYPE
    SYNTAX          Utf8String (SIZE(0..127))
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "Name of the application program source of the stream."
    ::=3D { rtpSenderEntry 6 }

rtpSenderSRs OBJECT-TYPE      =20
    SYNTAX          Counter32
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "A count of the number of RTCP Sender Reports that have=20
       been sent from this sender, or observed if the RTP entity
       is a monitor, since rtpSenderStartTime."
    ::=3D { rtpSenderEntry 7 }

rtpSenderSRTime OBJECT-TYPE
    SYNTAX          TimeStamp      =20
    MAX-ACCESS      read-only
    STATUS          current      =20
    DESCRIPTION
      "rtpSenderSRTime is the value of SysUpTime at the time that
       the last SR was received from this sender, in the case of a
       monitor or receiving host.  Or sent by this sender, in the=20
       case of a sending host."      =20
    ::=3D { rtpSenderEntry 8 }











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rtpSenderPT OBJECT-TYPE      =20
    SYNTAX          INTEGER (0..127)
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "Payload type from the RTP header of the most recently received=20
       RTP Packet (see RFC 1889, 'RTP: A Transport Protocol for=20
       Real-Time Applications' sec. 5)."      =20
    ::=3D { rtpSenderEntry 9 }

rtpSenderStartTime OBJECT-TYPE      =20
    SYNTAX          TimeStamp
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "The value of SysUpTime at the time that this row was=20
       created."      =20
    ::=3D { rtpSenderEntry 10 }

--
-- RECEIVER INVERSE TABLE
--
rtpRcvrInverseTable OBJECT-TYPE      =20
    SYNTAX          SEQUENCE OF RtpRcvrInverseEntry
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION
      "Maps rtpRcvrAddr and rtpSessionIndex to the rtpRcvrSRCSSRC and =20
       rtpRcvrSSRC indexes of the rtpRcvrTable.  This table allows=20
       management applications to find entries sorted by rtpRcvrAddr
       rather than by rtpSessionIndex. Given rtpSessionDomain and=20
       rtpRcvrAddr, a set of rtpSessionIndex, rtpRcvrSRCSSRC, and
       rtpRcvrSSRC values can be returned from a tree walk.  When=20
       rtpSessionIndex is specified in SNMP Get-Next operations, one or =

       more rtpRcvrSRCSSRC and rtpRcvrSSRC pairs may be returned."
    ::=3D { rtpMIBObjects 6 }












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rtpRcvrInverseEntry OBJECT-TYPE      =20
    SYNTAX          RtpRcvrInverseEntry
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION         =20
      "Each entry corresponds to exactly one entry in the
       rtpRcvrTable - the entry containing the index pair,
       rtpSessionIndex, rtpRcvrSSRC."
    INDEX { rtpSessionDomain, rtpRcvrAddr,  rtpSessionIndex,=20
            rtpRcvrSRCSSRC, rtpRcvrSSRC }
    ::=3D { rtpRcvrInverseTable 1 }

RtpRcvrInverseEntry ::=3D SEQUENCE { =20
        rtpRcvrInverseStartTime     TimeStamp     =20
        }

rtpRcvrInverseStartTime OBJECT-TYPE      =20
    SYNTAX          TimeStamp
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "The value of SysUpTime at the time that this row was=20
       created."      =20
    ::=3D { rtpRcvrInverseEntry 1 }

--
--  RECEIVERS TABLE
--
rtpRcvrTable OBJECT-TYPE      =20
    SYNTAX          SEQUENCE OF RtpRcvrEntry
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION
      "Table of information about a receiver or receivers of RTP=20
       session data. RTP hosts that receive RTP session packets=20
       MUST create an entry in this table for that receiver/sender=20
       pair.  RTP hosts that send RTP session packets MAY create
       an entry in this table for each receiver to their stream
       using RTCP feedback from the RTP group.  RTP monitors=20
       create an entry for each observed RTP session receiver as
       a side effect when a conceptual row in the rtpSessionTable
       is made 'active' by a manager."
    ::=3D { rtpMIBObjects 7 }






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rtpRcvrEntry OBJECT-TYPE      =20
    SYNTAX          RtpRcvrEntry
    MAX-ACCESS      not-accessible      =20
    STATUS          current
    DESCRIPTION         =20
      "Each entry contains information from a single RTP=20
       Synchronization Source that is receiving packets from the=20
       sender identified by rtpRcvrSRCSSRC (SSRC, see RFC 1889,=20
       'RTP: A Transport Protocol for Real-Time Applications'=20
       sec.6).  The session is identified to the the SNMP entity by=20
       rtpSessionIndex.  Rows are removed by the RTP agent when=20
       a BYE is received from the sender or when the sender times=20
       out (see RFC 1889, Sec. 6.2.1) or when the rtpSessionEntry is=20
       deleted."
    INDEX { rtpSessionIndex, rtpRcvrSRCSSRC, rtpRcvrSSRC }
    ::=3D { rtpRcvrTable 1 }

RtpRcvrEntry ::=3D SEQUENCE {
        rtpRcvrSRCSSRC        Unsigned32,      =20
        rtpRcvrSSRC           Unsigned32,
        rtpRcvrCNAME          Utf8String,
        rtpRcvrAddr           TAddress,      =20
        rtpRcvrRTT            Gauge32,
        rtpRcvrLostPackets    Counter64,      =20
        rtpRcvrJitter         Gauge32,
        rtpRcvrTool           Utf8String,
        rtpRcvrRRs            Counter32,
        rtpRcvrRRTime         TimeStamp,      =20
        rtpRcvrPT             INTEGER,=20
        rtpRcvrPackets        Counter64,      =20
        rtpRcvrOctets         Counter64,
        rtpRcvrStartTime      TimeStamp      =20
        }

rtpRcvrSRCSSRC OBJECT-TYPE
    SYNTAX       Unsigned32      =20
    MAX-ACCESS   not-accessible
    STATUS       current      =20
    DESCRIPTION
      "The RTP SSRC, or synchronization source identifier of the
       sender.  The RTP session address plus an SSRC uniquely=20
       identify a sender or receiver of an RTP stream (see RFC=20
       1889, 'RTP:  A Transport Protocol for Real-Time=20
       Applications' sec.3)."      =20
    ::=3D { rtpRcvrEntry 1 }



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rtpRcvrSSRC OBJECT-TYPE      =20
    SYNTAX       Unsigned32
    MAX-ACCESS   not-accessible      =20
    STATUS       current      =20
    DESCRIPTION
      "The RTP SSRC, or synchronization source identifier of the
       receiver.  The RTP session address plus an SSRC uniquely=20
       identify a receiver of an RTP stream (see RFC 1889, 'RTP: =20
       A Transport Protocol for Real-Time Applications' sec.3)."      =20
    ::=3D { rtpRcvrEntry 2 }

rtpRcvrCNAME OBJECT-TYPE      =20
    SYNTAX       Utf8String
    MAX-ACCESS   read-only      =20
    STATUS       current      =20
    DESCRIPTION
      "The RTP canonical name of the receiver."      =20
    ::=3D { rtpRcvrEntry 3 }

rtpRcvrAddr OBJECT-TYPE      =20
    SYNTAX       TAddress      =20
    MAX-ACCESS   read-only
    STATUS       current      =20
    DESCRIPTION
      "The unicast transport address on which the receiver is receiving =
RTP=20
       packets.  In the case of an RTP Monitor this address is the =
local
       address of the receiver on which it is sending RTCP Receiver =
Report=20
       packet is received."
    ::=3D { rtpRcvrEntry 4 }

rtpRcvrRTT OBJECT-TYPE      =20
    SYNTAX       Gauge32
    MAX-ACCESS   read-only      =20
    STATUS       current      =20
    DESCRIPTION
      "The round trip time measurement taken by the source of the=20
       RTP stream based on the algorithm described on sec. 6 of=20
       RFC 1889, 'RTP: A Transport Protocol for Real-Time=20
       Applications.'  This algorithm can produce meaningful=20
       results when the RTP agent has the same clock as the stream=20
       sender (when the RTP monitor is also the sending host for the
       particular reciever).  Otherwise, the entity should return=20
       'noSuchInstance' in response to queries against rtpRcvrRTT."
    ::=3D { rtpRcvrEntry 5 }




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rtpRcvrLostPackets OBJECT-TYPE      =20
    SYNTAX          Counter64=20
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "A count of RTP  packets lost as observed by this receiver
       since rtpRcvrStartTime."      =20
    ::=3D { rtpRcvrEntry 6 }

rtpRcvrJitter OBJECT-TYPE      =20
    SYNTAX          Gauge32
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "An estimate of delay variation as observed by this=20
       receiver.  (see RFC 1889, 'RTP: A Transport Protocol=20
       for Real-Time Applications' sec.6.3.1 and A.8)."
    ::=3D { rtpRcvrEntry 7 }

rtpRcvrTool OBJECT-TYPE      =20
    SYNTAX          Utf8String (SIZE(0..127))
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "Name of the application program source of the stream."
    ::=3D { rtpRcvrEntry 8 }

rtpRcvrRRs OBJECT-TYPE      =20
    SYNTAX          Counter32
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "A count of the number of RTCP Receiver Reports that have=20
       been sent from this receiver, or observed if the RTP entity
       is a monitor, since rtpRcvrStartTime."
    ::=3D { rtpRcvrEntry 9 }












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rtpRcvrRRTime OBJECT-TYPE      =20
    SYNTAX         TimeStamp
    MAX-ACCESS     read-only      =20
    STATUS         current      =20
    DESCRIPTION
      "rtpRcvrRRTime is the value of SysUpTime at the time that the=20
       last RTCP Receiver Report was received from this receiver, in=20
       the case of a monitor or RR receiver (the RTP Sender).  It is=20
       the  value of SysUpTime at the time that the last RR was sent by =

       this receiver in the case of an RTP receiver sending the RR."
    ::=3D { rtpRcvrEntry 10 }

rtpRcvrPT OBJECT-TYPE
    SYNTAX          INTEGER (0..127)      =20
    MAX-ACCESS      read-only
    STATUS          current      =20
    DESCRIPTION
      "Static or dynamic payload type from the RTP header (see=20
       RFC 1889, 'RTP: A Transport Protocol for Real-Time=20
       Applications' sec. 5)."      =20
    ::=3D { rtpRcvrEntry 11 }

rtpRcvrPackets OBJECT-TYPE      =20
    SYNTAX          Counter64
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "Count of RTP packets received by this RTP host receiver=20
       since rtpRcvrStartTime."
    ::=3D { rtpRcvrEntry 12 }

rtpRcvrOctets OBJECT-TYPE
    SYNTAX          Counter64      =20
    MAX-ACCESS      read-only
    STATUS          current      =20
    DESCRIPTION
      "Count of RTP octets received by this receiving RTP host=20
       since rtpRcvrStartTime."
    ::=3D { rtpRcvrEntry 13 }










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rtpRcvrStartTime OBJECT-TYPE      =20
    SYNTAX          TimeStamp
    MAX-ACCESS      read-only      =20
    STATUS          current      =20
    DESCRIPTION
      "The value of SysUpTime at the time that this row was=20
       created."      =20
    ::=3D { rtpRcvrEntry 14 }








































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--
--  MODULE GROUPS
--
rtpGroups OBJECT IDENTIFIER ::=3D { rtpConformance 1 }
rtpSystemGroup      OBJECT-GROUP      =20
    OBJECTS         {
                    rtpSessionDomain,
                    rtpSessionRemAddr,
                    rtpSessionIfIndex,
                    rtpSessionSenderJoins,
                    rtpSessionReceiverJoins,
                    rtpSessionStartTime,
                    rtpSessionByes,                      =20
                    rtpSessionMonitor,
                    rtpSenderCNAME,                      =20
                    rtpSenderAddr,
                    rtpSenderPackets,                      =20
                    rtpSenderOctets,
                    rtpSenderTool,                      =20
                    rtpSenderSRs,
                    rtpSenderSRTime,
                    rtpSenderStartTime,                      =20
                    rtpRcvrCNAME,
                    rtpRcvrAddr,                      =20
                    rtpRcvrLostPackets,
                    rtpRcvrJitter,                      =20
                    rtpRcvrTool,
                    rtpRcvrRRs,                       =20
                    rtpRcvrRRTime,
                    rtpRcvrStartTime                      =20
                    }
    STATUS          current      =20
    DESCRIPTION     =20
        "Objects used by all RTP systems."      =20
    ::=3D { rtpGroups 1 }














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rtpHostGroup    OBJECT-GROUP      =20
    OBJECTS     {
                rtpSessionLocAddr,                      =20
                rtpSenderPT,
                rtpRcvrPT,                      =20
                rtpRcvrRTT,
                rtpRcvrOctets,                      =20
                rtpRcvrPackets
                }      =20
    STATUS      current      =20
    DESCRIPTION     =20
           "Objects used by all RTP Host systems and optionally by=20
            RTP Monitor systems."      =20
    ::=3D { rtpGroups 2 }

rtpMonitorGroup	OBJECT-GROUP      =20
    OBJECTS     {
                rtpSessionNewIndex,                      =20
                rtpSessionRowStatus
                }
    STATUS      current      =20
    DESCRIPTION
        "Objects used by RTP monitor systems."      =20
    ::=3D { rtpGroups 3 }

rtpInverseGroup OBJECT-GROUP
    OBJECTS     {
                rtpSessionInverseStartTime,
                rtpSenderInverseStartTime,
                rtpRcvrInverseStartTime
				}
    STATUS      current
    DESCRIPTION
	    "Objects used in the Inverse Lookup Tables."
    ::=3D { rtpGroups 4 }










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--
--  Compliance
--
rtpCompliances OBJECT IDENTIFIER ::=3D { rtpConformance 2 }

rtpHostCompliance MODULE-COMPLIANCE      =20
    STATUS          current
    DESCRIPTION              =20
	    "Host implementations must comply."
    MODULE           RTP-MIB         =20
    MANDATORY-GROUPS {
                     rtpSystemGroup,
                     rtpHostGroup
    				 }
    GROUP            rtpMonitorGroup
    DESCRIPTION
        "The objects in the rtpHostGroup MUST be implemented=20
         in RTP host systems, i.e., that are the source or the=20
         destination of RTP data packets."
    GROUP            rtpInverseGroup
    DESCRIPTION
        "Multicast RTP Systems SHOULD implement the optional
         tables."
        OBJECT  rtpSessionNewIndex
            MIN-ACCESS not-accessible             =20
                DESCRIPTION
                 "RTP system implementations support of
                  row creation and deletion is OPTIONAL so
                  implementation of this object is OPTIONAL."
        OBJECT  rtpSessionDomain            =20
           MIN-ACCESS read-only
                DESCRIPTION
                 "RTP system implementation support of
                  row creation and deletion is OPTIONAL.  When
                  it is not supported so write access is=20
                  OPTIONAL."         =20
        OBJECT  rtpSessionRemAddr
            MIN-ACCESS read-only             =20
              DESCRIPTION
               "Row creation and deletion is OPTIONAL so
                read-create access to this object is OPTIONAL."
     	OBJECT  rtpSessionIfIndex            =20
     	    MIN-ACCESS read-only
              DESCRIPTION
               "Row creation and deletion is OPTIONAL so
                read-create access to this object is OPTIONAL."



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        OBJECT  rtpSessionRowStatus            =20
            MIN-ACCESS not-accessible
              DESCRIPTION
               "Row creation and deletion is OPTIONAL so
                read-create access to this object is OPTIONAL."
        OBJECT  rtpSessionInverseStartTime
            MIN-ACCESS not-accessible
              DESCRIPTION
               "Multicast RTP Systems SHOULD implement the optional
                tables."
        OBJECT  rtpSenderInverseStartTime
            MIN-ACCESS not-accessible
              DESCRIPTION
               "Multicast RTP Systems SHOULD implement the optional
                tables."
        OBJECT  rtpRcvrInverseStartTime
            MIN-ACCESS not-accessible
              DESCRIPTION
               "Multicast RTP Systems SHOULD implement the optional
                tables."
    ::=3D { rtpCompliances 1 }




























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rtpMonitorCompliance MODULE-COMPLIANCE      =20
    STATUS          current
    DESCRIPTION              =20
	  "Monitor implementations must comply.  RTP Monitors are not=20
	  required to support creation or deletion."
    MODULE           RTP-MIB         =20
    MANDATORY-GROUPS     {=20
                         rtpSystemGroup,
                         rtpMonitorGroup
                         }
    GROUP                rtpHostGroup
    DESCRIPTION
        "The objects in the rtpMonitorGroup MUST be
         implemented in RTP monitors."         =20
    GROUP                rtpInverseGroup
    DESCRIPTION
        "Multicast RTP Systems SHOULD implement the optional
         tables."
        OBJECT  rtpSessionLocAddr            =20
            MIN-ACCESS not-accessible
              DESCRIPTION
               "RTP monitor sourcing of RTP or RTCP data packets=20
                is OPTIONAL and implementation of this object is=20
                OPTIONAL."
        OBJECT  rtpRcvrPT
            MIN-ACCESS not-accessible             =20
              DESCRIPTION
               "RTP monitor systems NEED NOT support
                retrieval of the RTP Payload Type from the RTP=20
                header (and may receive RTCP messages only).  When
                queried for the payload type information"
      	OBJECT  rtpSenderPT            =20
            MIN-ACCESS not-accessible
              DESCRIPTION
               "RTP monitor systems NEED NOT support
                retrieval of the RTP Payload Type from the RTP=20
                header (and may receive RTCP messages only).  When
                queried for the payload type information."
        OBJECT  rtpRcvrOctets
            MIN-ACCESS not-accessible             =20
              DESCRIPTION
               "RTP monitor systems may receive only the RTCP messages
                and not the RTP messages that contain the octet count
                of the RTP message.  Thus implementation of this
                object is OPTIONAL"          =20




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Internet Draft  RTP MIB                                    October 1999




        OBJECT  rtpRcvrPackets
            MIN-ACCESS not-accessible             =20
              DESCRIPTION
               "RTP monitor systems may receive only the RTCP messages
                and not the RTP messages that contain the octet count
                of the RTP message.  Thus implementation of this
                object is OPTIONAL."       =20
     	OBJECT  rtpSessionIfIndex            =20
     	    MIN-ACCESS read-only
              DESCRIPTION
               "Row creation and deletion is OPTIONAL so
                read-create access to this object is OPTIONAL."
        OBJECT  rtpSessionInverseStartTime
            MIN-ACCESS not-accessible
              DESCRIPTION
               "Multicast RTP Systems SHOULD implement the optional
                tables."
        OBJECT  rtpSenderInverseStartTime
            MIN-ACCESS not-accessible
              DESCRIPTION
               "Multicast RTP Systems SHOULD implement the optional
                tables."
        OBJECT  rtpRcvrInverseStartTime
            MIN-ACCESS not-accessible
              DESCRIPTION
               "Multicast RTP Systems SHOULD implement the optional
                tables."
    ::=3D { rtpCompliances 2 }
END



















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Internet Draft  RTP MIB                                    October 1999



4.  Security Issues

In most cases, MIBs are not themselves security risks; if SNMP security
is operating as intended, the use of a MIB to view information about a=20
system, or to change some parameter at the system, is a tool, not a=20
threat.  However, there are a number of management objects defined in =
this
MIB that have a MAX-ACCESS clause of read-write and/or read-create.  Suc=
h
objects may be considered sensitive or vulnerable in some network
environments.  The support for SET operations in a non-secure
environment without proper protection can have a negative effect on
network operations.

None of the read-only objects in this MIB reports a password, though
some SDES [RFC1889] items such as the CNAME [RFC1889], the canonical =
name,=20
may be deemed sensitive depending on the security policies of a =
particular=20
enterprise.  If access to these objects is not limited by an=20
appropriate access control policy, these objects can provide an=20
attacker with information about a system's configuration and the=20
services that that system is providing.  Some enterprises view their=20
network and system configurations, as well as information about=20
usage and performance, as corporate assets; such enterprises may=20
wish to restrict SNMP access to most of the objects in the MIB.
This MIB supports read-write operations against rtpSessionNewIndex=20
which has the side effect of creating an entry in the rtpSessionTable=20
when it is written to.  Five objects in rtpSessionEntry have=20
read-create access: rtpSessionDomain, rtpSessionRemAddr,=20
rtpSessionIfIndex, rtpSessionRowStatus, and rtpSessionIfAddr identify =
an RTP
session to be monitored on a particular interface.  The values of these =

objects are not to be changed once created, and initialization of these =

objects affects only the monitoring of an RTP session and not the=20
operation of an RTP session on any host end-system.  Since write=20
operations to rtpSessionNewIndex and the five objects in=20
rtpSessionEntry affect the operation of the monitor, write access to=20
these objects should be subject to the appropriate access control=20
policy.

Confidentiality of RTP and RTCP data packets is defined in section 9 of =

the RTP specification [RFC1889].  Encryption may be performed on RTP =
packets,
RTCP packets, or both.  Encryption of RTCP packets may pose a problem
for third-party monitors though "For RTCP, it is allowed to split a
compound RTCP packet into two lower-layer packets, one to be encrypted
and one to be sent in the clear.  For example, SDES information might
be encrypted while reception reports were sent in the clear to=20
accommodate third-party monitors [RFC1889]." =20





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SNMPv1 by itself is not a secure environment.  Even if the network
itself is secure (for example by using IPSec), even then, there is no
control as to who on the secure network is allowed to access and
GET/SET (read/change/create/delete) the objects in this MIB.
It is recommended that the implementers consider the security
features as provided by the SNMPv3 framework.  Specifically, the use
of the User-based Security Model RFC 2574 [RFC2574] and the View-
based Access Control Model RFC 2575 [RFC2575] is recommended.
It is then a customer/user responsibility to ensure that the SNMP
entity giving access to an instance of this MIB, is properly
configured to give access to the objects only to those principals
(users) that have legitimate rights to indeed GET or SET
(change/create/delete) them.

5.  Acknowledgements

The authors wish to thank Bert Wijnen and the participants from the=20
ITU SG-16 management effort for their helpful comments.  Alan Batie
and Bill Lewis from Intel also contributed greatly to the RTP MIB
through their review of various drafts of the MIB and their work
on the implementation of an SNMP RTP Monitor.  Stan Naudus from 3Com=20
and John Du from Intel contributed to the original RTP MIB design and=20
co-authored the original RTP MIB draft documents; much of their work=20
remains in the current RTP MIB.

6.  Intellectual Property

The IETF takes no position regarding the validity or scope of any
intellectual property or other rights that might be claimed to
pertain to the implementation or use of the technology described in
this document or the extent to which any license under such rights
might or might not be available; neither does it represent that it
has made any effort to identify any such rights.  Information on the
IETF's procedures with respect to rights in standards-track and
standards-related documentation can be found in BCP-11.  Copies of
claims of rights made available for publication and any assurances of
licenses to be made available, or the result of an attempt made to
obtain a general license or permission for the use of such
proprietary rights by implementors or users of this specification can
be obtained from the IETF Secretariat.

The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights which may cover technology that may be required to practice
this standard.  Please address the information to the IETF Executive
Director.



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7.  References

[RFC1889]   H. Shulzrinne, S. Casner, R. Frederick, and V. Jacobson, =
"RTP:
            A Transport Protocol for real-time applications," RFC 1889.

[RFC2571]   Harrington, D., Presuhn, R., and B. Wijnen, "An =
Architecture
            for Describing SNMP Management Frameworks", RFC 2571, April
            1999

[RFC1155]   Rose, M., and K. McCloghrie, "Structure and Identification
            of Management Information for TCP/IP-based Internets", STD
            16, RFC 1155, May 1990

[RFC1212]   Rose, M., and K. McCloghrie, "Concise MIB Definitions", STD
            16, RFC 1212, March 1991

[RFC1215]   M. Rose, "A Convention for Defining Traps for use with the
            SNMP", RFC 1215, March 1991

[RFC2578]   McCloghrie, K., Perkins, D., Schoenwaelder, J., Case, J.,
            Rose, M., and S. Waldbusser, "Structure of Management
            Information Version 2 (SMIv2)", STD 58, RFC 2578, April =
1999

[RFC2579]   McCloghrie, K., Perkins, D., Schoenwaelder, J., Case, J.,
            Rose, M., and S. Waldbusser, "Textual Conventions for
            SMIv2", STD 58, RFC 2579, April 1999

[RFC2580]   McCloghrie, K., Perkins, D., Schoenwaelder, J., Case, J.,
            Rose, M., and S. Waldbusser, "Conformance Statements for
            SMIv2", STD 58, RFC 2580, April 1999

[RFC1157]   Case, J., Fedor, M., Schoffstall, M., and J. Davin, "Simple
            Network Management Protocol", STD 15, RFC 1157, May 1990.

[RFC1901]   Case, J., McCloghrie, K., Rose, M., and S. Waldbusser,
            "Introduction to Community-based SNMPv2", RFC 1901, January
            1996.

[RFC1906]   Case, J., McCloghrie, K., Rose, M., and S. Waldbusser,
            "Transport Mappings for Version 2 of the Simple Network
            Management Protocol (SNMPv2)", RFC 1906, January 1996.

[RFC2572]   Case, J., Harrington D., Presuhn R., and B. Wijnen, =
"Message
            Processing and Dispatching for the Simple Network =
Management
            Protocol (SNMP)", RFC 2572, April 1999
           =20



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[RFC2574]   Blumenthal, U., and B. Wijnen, "User-based Security Model
            (USM) for version 3 of the Simple Network Management
            Protocol (SNMPv3)", RFC 2574, April 1999

[RFC1905]   Case, J., McCloghrie, K., Rose, M., and S. Waldbusser,
            "Protocol Operations for Version 2 of the Simple Network
            Management Protocol (SNMPv2)", RFC 1905, January 1996.

[RFC2573]   Levi, D., Meyer, P., and B. Stewart, "SNMPv3 Applications",
            RFC 2573, April 1999

[RFC2575]   Wijnen, B., Presuhn, R., and K. McCloghrie, "View-based
            Access Control Model (VACM) for the Simple Network
            Management Protocol (SNMP)", RFC 2575, April 1999

[RFC2570]   Case, J., Mundy, R., Partain, D., and B. Stewart,
            "Introduction to Version 3 of the Internet-standard Network
            Management Framework", RFC 2570, April 1999































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Internet Draft  RTP MIB                                    October 1999



8. Authors Addresses

Mark Baugher                Email: mbaugher@intel.com
Intel Corporation
2111 N.E.25th Avenue =20
Hillsboro, Oregon  97124
U.S.A.
            =20
Bill Strahm                 Email: Bill.Strahm@intel.com  =20
Intel Corporation
2111 N.E.25th Avenue =20
Hillsboro, Oregon  97124
U.S.A.
                       =20
Irina Suconick              Email: isuconick@videoserver.com
Videoserver Corporation
63 Third Street
Burlington, Massachusetts  01803
U.S.A.

9. Full Copyright Statement

   Copyright (C) The Internet Society (1999). All Rights Reserved.
  =20
   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph=20
   are included on all such copies and derivative works.  However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the  purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.
  =20
   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.
  =20
   This document and the information contained herein is provided on an
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
  =20

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------_=_NextPart_000_01BF1CBB.FACE2E27--



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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "Internet drafts" <internet-drafts@ietf.org>,
        "IETF-avt list" <rem-conf@es.net>
Cc: "Michele Mizzaro" <michele.mizzaro@ausys.se>,
        "Mickey Nasiri (ECS)" <mickey.nasiri@ecs.ericsson.se>
Subject: New version of draft-ietf-avt-rtp-text
Date: Fri, 22 Oct 1999 22:13:54 +0200
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Please find the attached new revision of the RTP-text specification
to be reviewed in the Washington IETF / avt meeting.

File name: draft-ietf-avt-rtp-text-02.txt

Main changes:
Stricting up nomenclature.
Send empty packets to squeeze out redundant packets.
Specify order of redundant data for recovery procedure to work.

Regards
Gunnar Hellstr=F6m

______________________________________
Gunnar Hellstr=F6m
LM Ericsson
Sweden

Mob +46 708 204 288
Tel +46 8 556 002 03
Fax +46 8 556 002 06
Video +468 556 002 05
Txt (All kinds) +46 8 556 002 05
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Internet Engineering Task Force                                   AVT WG
Internet Draft                                          Gunnar Hellstrom
draft-ietf-avt-rtp-text-02.txt                              LM  Ericsson
October  22, 1999
Expires: April 22, 2000


                      RTP Payload for Text Conversation

STATUS OF THIS MEMO

   This document is an Internet-Draft and it is in full conformance with
   all provisions of section 10 of RFC2026. Internet-Drafts are working
   documents of the Internet Engineering Task Force (IETF), its areas,
   and its working groups.  Note that other groups may also distribute
   working documents as Internet-Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as ``work in progress''.

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/lid-abstracts.txt

   The list of Internet-Draft Shadow Directories can be accessed at 
   http://www.ietf.org/shadow.html

   Distribution of this document is unlimited.


ABSTRACT

   This memo describes how to carry text conversation session contents
   in RTP packets. Text conversation session contents are specified in
   ITU-T Recommendation T.140 [1]. 

   Text conversation is used alone or in connection to other
   conversational facilities such as video and voice, to form multimedia
   conversation services.

   This RTP payload description contains an optional possibility to
   include redundant text from already transmitted packets in order to
   reduce the risk of text loss caused by packet loss. The redundancy
   coding follows RFC 2198.










Hellstrom                                                       [Page 1]


Internet Draft


1 Introduction

   This memo defines a payload type for carrying text conversation
   session contents in RTP packets. Text conversation session contents
   are specified in ITU-T Recommendation T.140 [1]. Text conversation is
   used alone or in connection to other conversational facilities such
   as video and voice, to form multimedia conversation services. Text in
   text conversation sessions is sent as soon as it is available, or 
   with a small delay for buffering.
   The text is supposed to be entered by human users from a keyboard,
   handwriting recognition, voice recognition or any other input method.
   The rate of character entry is usually at a level of a few characters
   per second or less. Therefore, the expected number of characters to
   transmit is low. Only one or a few new characters are expected to be
   transmitted with each packet.

   T.140 specifies that text and other T.140 elements SHALL be
   transmitted in ISO 10 646-1 code with UTF-8 transformation. That
   makes it easy to implement internationally useful applications, and
   to handle the text in modern information technology environments.
   The payload of an RTP packet following this specification
   consists of text encoded according to T.140 without any
   additional framing.  A common case will be a single ISO 10646
   character, UTF-8 encoded.
 
   T.140 requires the transport channel to provide characters without
   duplication and in original order. 
   Text conversation users expect that text will be delivered with no or
   a low level of lost information. If lost information can be
   indicated, the willingness to accept loss is expected to
   be higher.
 
   Therefore a mechanism based on RTP is specified here. It gives text
   arrival in correct order, without duplications, 
   and with detection and indication of losses.  It also includes an
   optional possibility to repeat data for redundancy to lower the
   risk of loss. Since packet overhead is
   usually much larger than the T.140 contents, the increase in channel
   load by the redundancy scheme is minimal. 



1.1 Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [4]




Hellstrom                                                       [Page 2]


Internet Draft                  			

2. Usage of RTP

   When transport of T.140 text session data in RTP is desired, the
   payload as described in this specification SHOULD be used.

   A text conversation RTP packet as specified by this payload format
   consists of an RTP header as defined in RFC 1889 [2] followed
   immediately by a block of T.140 data, defined here to be a
   "T140block".  There is no additional header specific to this
   payload format.  The T140block contains one or more T.140 code
   elements as specified in [1].  Most T.140 code elements are single
   ISO 10646 [5] characters, but some are multiple character
   sequences.  Each character is UTF-8 encoded [6] into one or more
   octets.

   The T140blocks MAY be transmitted redundantly according to the
   payload format defined in RFC 2198 [3].  In that case, the RTP
   header is followed by one or more redundant data block headers, the
   same number of redundant data fields carrying T140blocks from
   previous packets, and finally the new (primary) T140block for this
   packet.



2.1 RTP packet header
   
   Each RTP packet starts with a fixed RTP header. The following fields
   of the RTP fixed header are used for T.140 text streams:

Payload Type (PT): The assignment of an RTP payload type is specific
   to the RTP profile under which this payload format is used.  For
   profiles which use dynamic payload type number assignment, this
   payload format is identified by the name "T140" (see section 6).
   If redundancy is used per RFC 2198, the Payload Type SHALL indicate
   that payload format ("RED").

Sequence number:  The Sequence Number SHALL be increased by one for
   each new transmitted packet. It is used for detection of packet loss
   and packets out of order, and can be used in the process of retrieval
   of redundant text, reordering of text and marking missing text.

Timestamp: The RTP Timestamp encodes the approximate sampling instance
   of the primary text in the packet. A clock frequency of 1000 Hz SHALL
   be used. Sequential packets MUST NOT use the same timestamp because
   the timestamps are used for sequencing of redundant T140blocks.






   
Hellstrom                                                       [Page 3]


Internet Draft                                  

2.2 Additional headers

   There are no additional headers defined specific to this payload
   format.

   When redundant transmission of the data according to RFC 2198 is
   desired, the RTP header is followed by one or more redundant data
   block headers, one for each redundant data block to be included.
   Each of these headers provides the timestamp offset and length of
   the corresponding data block plus a payload type number indicating
   this payload format ("T140").

2.3 T.140 Text structure

   T.140 text is UTF-8 coded as specified in T.140 with no extra
   framing. When using the format with redundant data, the transmitter
   MAY select a number of T140block generations to retransmit in each
   packet. A higher number introduces better protection against loss
   of text but increases the data rate.

   Because the redundant data block headers do not contain sequence
   number information, some rules must be followed to allow the
   redundant data corresponding to missing primary to be merged
   properly into the stream of primary data T140blocks:
      - Each redundant data block MUST contain the same data as a
        T140block previously transmitted as primary data, and be
        identified with a timestamp offset equating to the original
        timestamp for that T140block.
      - The redundant data MUST be placed in age order with most
        recent redundant T140block last in the redundancy area.
      - All T140blocks from the oldest desired generation up through
        the generation immediately preceding the new (primary)
        T140block MUST be included.

   These rules allow the sequence numbers for the redundant T140blocks
   to be inferred by counting backwards from the sequence number in
   the RTP header.  The result will be that all the text in the
   payload will be contiguous and in order.


3. Recommended procedures

   This section contains RECOMMENDED procedures for usage of the payload
   format.
   Based on the information in the received packets, the receiver can:
- reorder received text out of order.
      - mark where text is missing because of packet loss.
      - compensate for lost packets by using redundant data.




Hellstrom                                                       [Page 4]


Internet Draft                  			


3.1 Recommended basic procedure   

   On reception, the RTP sequence number is compared with the sequence
   number of the last correctly received packet. If they are 
   consecutive, the T140block is retrieved from the packet.  If 
   redundant transmission is in use, any redundant data blocks in the 
   received packet are ignored and only the new (primary) T140block is
   retrieved.

3.2 Recommended procedure for compensation for lost packets.

   For reduction of data loss in case of packet loss, redundant data MAY
   be included in the packets following to the procedures in RFC 2198.
   If network conditions are not known, it is RECOMMENDED to use one
   redundant T140block in each packet. If there is a gap in
   the RTP sequence numbers, and redundant T140blocks are
   available in a subsequent packet, the sequence numbers for the
   redundant T140blocks should be inferred by counting backwards from
   the sequence number in the RTP header for that packet.  If there are
   redundant T140blocks with sequence numbers matching those that are
   missing, the redundant T140blocks may be substituted for the
   missing T140blocks.

   Both for the case when redundancy is used and not used, missing data
   SHOULD be marked. T.140 defines a missing data marker (Unicode
   character 2607 "Lightning") which SHOULD be used as a marker to
   indicate each missing T140block.  



3.3 Recommended procedure for compensation for packets out of order.

   For protection against packets arriving out of order, the following
   procedure MAY be implemented in the receiver.
   If analysis of a received packet reveals a gap in the sequence and
   no redundant data is available to fill that gap,
   the received packet can be kept in a buffer to allow time for the
   missing packet(s) to arrive.  It is suggested that the waiting
   time be limited to 0.5 second or until three packets subsequent to
   the missing packet(s) have arrived, whichever is shorter.
   If a packet with a T140block belonging to
   the gap arrives before the waiting time expires, this T140block
   is inserted into the gap and then consecutive T140blocks from the
   leading edge of the gap may be consumed.  Any T140block which does
   not arrive before the time limit expires should be treated as lost.






Hellstrom                                                       [Page 5]


Internet Draft                                  




3.4 Transmission during "silent periods" when redundancy is used.

   When using the redundancy transmission scheme, and there is nothing
   more to transmit from T.140, the latest T140block has a risk of
   getting old before it is transmitted as redundant data. The result
   is less useful protection against packet loss at the end of a text
   input sequence. For cases where this should be avoided,
   a zero-length primary T140block MAY be transmitted with the redundant
   data.  

   Any zero-length T140blocks that are sent as primary data
   SHOULD be included as redundant T140blocks on subsequent packets
   just as normal text T140blocks would be so that sequence number
   inference for the redundant T140blocks will be correct, as
   explained in section 2.3.

   Redundancy for the last T140block SHOULD NOT be implemented by
   repeatedly transmitting the same packet (with the same sequence
   number) because this will cause the packet loss count, as reported
   in RTCP, to decrement.





























Hellstrom                                                       [Page 6]


Internet Draft                                  



4. Examples

   This is an example of a T140 RTP packet without redundancy.
    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=2|P|X| CC=0  |M|   T140 PT   |       sequence number         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                      timestamp (1000Hz)                       |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   +                      T.140 encoded data                       +
   |                                                               |
   +                                               +---------------+
   |                                               |               
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               

    This is an example of an RTP packet with one redundant T140block.
    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=2|P|X| CC=0  |M|  "RED" PT   |   sequence number of primary  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              timestamp  of primary encoding "P"               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |1|   T140 PT   |  timestamp offset of "R"  | "R" block length  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |0|   T140 PT   |                                               |
   +-+-+-+-+-+-+-+-+                                               +
   |                                                               |
   +               "R" T.140 encoded redundant data                +
   |                                                               |
   +                                               +---------------+
   |                                               |               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               +
   |                "P" T.140 encoded primary data                 |
   +                                                               +
   +                                               +---------------+
   |                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Figure: Examples of RTP text packets.





Hellstrom                                                       [Page 7]


Internet Draft

5.  Security Considerations.

    Since the intention of the described payload format is to carry text
    in a text conversation, security measures in the form of encryption
    are of importance. The amount of data in a text conversation session
    is low and therefore any encryption method MAY be selected and
    applied to T.140 session contents or to the whole RTP packets. 
    When redundant data is included, the same security considerations
    as for RFC 2198 apply.

6  MIME Media Type Registrations

    This document defines a new RTP payload name and associated MIME
    type, T140 (text/t140). The registration form for this MIME type has
    been submitted to IANA.
   
6.1  Registration of MIME media type text/t140

     MIME media type name: text

     MIME subtype name: t140

     Required parameters: None

     Optional parameters: None

     Encoding considerations: T140 text can be transmitted 
     with RTP as specified in "draft-ietf-avt-rtp-text-02".

     Security considerations: None

     Interoperability considerations: None

     Published specification: ITU-T T.140 Recommendation.
                              draft-ietf-avt-rtp-text-02.     

     Applications which use this media type:
       Text communication terminals and text conferencing tools.

     Additional information: None

       Magic number(s): None
       File extension(s): None
       Macintosh File Type Code(s): None

     Person & email address to contact for further information:
       Gunnar Hellstrom
       e-mail: gunnar.hellstrom@omnitor.se

     Intended usage: COMMON
     Author/Change controller:
       Gunnar Hellstrom
       e-mail: gunnar.hellstrom@omnitor.se
Hellstrom                                                       [Page 8]

Internet Draft                                Expires: April 22, 2000


7.  Author's Address

   Gunnar Hellstrom
   Ericsson Mobile Communication AB
   Home Communications
   SE-164 80 Stockholm   
   Sweden

   e-mail: gunnar.hellstrom@omnitor.se
   Tel:    +46 708 204 288
   Fax:    +46 8 556 002 06

8 References

[1] ITU-T Recommendation T.140 (1998) - Text conversation protocol for
    multimedia application, with amendment 1999.

[2] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, 
   "RTP: A Transport Protocol for Real-Time Applications", RFC 1889.

[3] C. Perkins, I. Kouvelas, V. Hardman, M. Handley, and J. Bolot,
   "RTP payload for redundant audio data," RFC 2198, Internet
   Engineering Task Force, Sept. 1997.

[4] S. Bradner, "Key words for use in RFCs to Indicate Requirement
    Levels", RFC 2119, March 1997.

[5] ISO/IEC 10646-1: (1993), Universal Multiple Octet Coded Character
    Set

[6] F. Yergeau, "UTF-8, a transformation format of ISO 10646,"  RFC
    2279, Internet Engineering Task Force, Jan. 1998.

9   Revision history.

    Oct 6 1999. draft-01. Explanations in recommended procedures
                          improved.
    Oct 22 1999. draft-02. Nomenclature, ordering of redundant packets, 
                           send empty info to expire redundancy.











Hellstrom                                                       [Page 9]



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Date: Fri, 22 Oct 1999 13:09:59 -0700
To: rem-conf@es.net
From: Dave Singer <singer@apple.com>
Subject: Updated hint track format
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The MPEG folks have asked that we update the hint track format 
specification for RTP hint tracks in QuickTime files to both bodies. 
I have sent this in to the IETF but it seems to be having trouble 
getting through.  It is provided as an informative and discussion 
document at this point.

I'll be available to discuss it in Washington, if appropriate.
--============_-1271499387==_============
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--============_-1271499387==_============
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David Singer
Apple Computer/QuickTime
--============_-1271499387==_============--



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Message-ID: <016301bf1ccd$dd063c30$df82cf87@research.att.com>
From: "Baldine Paul" <baldine@research.att.com>
To: <cabo@tzi.org>, <lscline@jf.intel.com>, <thomas.r.gardos@intel.com>,
        "Gary Sullivan" <garys@pictel.com>, <stewe@cs.tu-berlin.de>
Cc: <rem-conf@es.net>
Subject: H.263 RFC 2429
Date: Fri, 22 Oct 1999 16:41:45 -0400
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I have a comment regarding the packetization scheme described in=20
Section 5.1.2 of the RFC. It specifies that a redundant copy of the =
picture
header may be attached to the RTP header for packets that begin with =
GBSC or SSC.
What is not obvious is whether that redundant picture header is full =
length (with UFEP=3D001) or
not.=20
Although, it could be left to the implementer to decide whether to =
include a complete picture
header or not, I suggest, to be consistent with the recommendation of =
Section 5.1.1, that
all redundant picture headers(for packets beginning with a Picture Start =
Code,
a GBSC, a SSC or even a Follow-on packet), be complete, with UFEP=3D001.
=20
Baldine-Brunel Paul
AT&T Labs-Research
Room 3-225, 100 Schultz Drive
Red Bank, NJ 07701
phone: 732-345-3312
baldine@research.att.com
http://www.research.att.com/info/baldine


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	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2614.3401" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I have a comment regarding the =
packetization scheme=20
described in </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Section 5.1.2 of the RFC. It specifies =
that a=20
redundant copy of the picture</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>header may be attached to the RTP =
header for=20
packets that begin with GBSC or SSC.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>What is not obvious is whether that =
redundant=20
picture header is full length (with UFEP=3D001) or</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>not. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Although, it could be left to the =
implementer to=20
decide whether to include a complete picture</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>header or not, I suggest, to be =
consistent with the=20
recommendation of Section 5.1.1, that</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>all redundant picture headers(for =
packets beginning=20
with a Picture Start Code,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>a GBSC, a SSC or even a Follow-on=20
packet),</FONT><FONT face=3DArial size=3D2> be complete, with =
UFEP=3D001.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Baldine-Brunel Paul<BR>AT&amp;T=20
Labs-Research<BR>Room 3-225, 100 Schultz Drive<BR>Red Bank, NJ =
07701<BR>phone:=20
732-345-3312<BR><A=20
href=3D"mailto:baldine@research.att.com">baldine@research.att.com</A><BR>=
<A=20
href=3D"http://www.research.att.com/info/baldine">http://www.research.att=
.com/info/baldine</A></FONT></DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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From rem-conf-request@es.net  Sat Oct 23 00:08:31 1999
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Message-ID: <01af01bf1d07$be70bfe0$1cc77885@pcg-c1r>
From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: <rem-conf@es.net>, "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>
Cc: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
Subject: Submission of I/D: RTP payload format for MPEG-4 Audio/Visual streams
Date: Sat, 23 Oct 1999 12:33:43 +0900
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Dear all,

I submitted an Internet draft for the RTP payload format for MPEG-4
Audio/Visual streams:

Title : RTP payload format for MPEG-4 Audio/Visual streams
Author(s) : Yoshihiro Kikuchi, Toshiyuki Nomura, Shigeru Fukunaga,
                 Yoshinori Matsui, Hideaki Kimata
Filename : draft-jnb-mpeg4av-rtp-00.txt
Pages : 14
Date : 22-Oct-99

This document describes RTP payload formats for the carriage of MPEG-4
Audio and Visual streams, and an RTCP format for MPEG-4 backward channel
messages functionalities. The specification is based on the study on
Internet draft for the carriage of MPEG-4 on IP, which is sent from
ISO/IEC JTC1/SC29/WG11(MPEG) as a liaison statement. In this
specification, MPEG-4 Audio/Visual bitstreams are directly mapped into
RTP packets without using MPEG-4 Systems. The RTP header fields usage
and the fragmentation rule for MPEG-4 Visual and Audio bitstreams are
specified. It also specifies an RTCP packet usage to carry the MPEG-4
backward channel messages.



I attach this document. Review and comments by the experts are much
appreciated.

Best regards,
Yoshihiro Kikuchi

----
        Yoshihiro Kikuchi

E-mail: kiku@eel.rdc.toshiba.co.jp
Corporate Research and Development Center
TOSHIBA Corporation
TEL: +81 +44 549 2288    FAX: +81 +44 520 1308





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	id 11ez1K-0000so-00; Sat, 23 Oct 1999 04:07:36 -0700
From: The Bedrisers!<rogerdoer@bigfoot.com>
To: relyea@msn.com
Subject: Picture a BRAND NEW LOOK for every bedroom in your home!
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<base href="http://www.bedriser.com/">
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">

<html>
<head>
	<title>BedRiser Blowout Special!!</title>
</head>

<body bgcolor="white">
<font face="Arial">
Click the following to enlarge page:
<a href="http://www.bedriser.com/brspecial.htm">http://www.bedriser.com</a><br>
If you do not wish to receive any mailings from us, click
<a href="mailto:bedriser@bedriser.com?subject=remove">HERE</a> to be removed.
<TABLE width="100%" border=1>
  <TBODY>
  
  <TR>
    <TD colSpan=3>
      <P align=center><FONT size=5><STRONG>
      Breakthrough Decorating &amp; Storage Product<br>
      Guaranteed To Make Your Bedroom Beautiful 
      &amp; Cozy</STRONG></FONT></P>
      <P align=center>&nbsp;</P>
             </TD></TR>
  <TR>
    <TD><br><FONT 
      color=#0000ff><STRONG><i>Discover The Benefits Of The BedRiser:</i><br></STRONG></FONT><ul><li width='"5%"'>Ends <FONT color=#ff0099><STRONG><u>back pain</u> 
        </STRONG></FONT>   from bending to make the bed.
		<br><IMG src="images/bspace.gif"><li>Creates tons of <FONT color=#ff0099><STRONG><u>hidden storage</u></STRONG></FONT>.<br><IMG src="images/bspace.gif"><li>New height turns your bed into the <FONT 
        color=#ff0099><STRONG><u>ultimate laundry table</u></STRONG></FONT>.<br><IMG src="images/bspace.gif"><li><FONT color=#ff0099><STRONG><u>Back 
        problems?</u></STRONG> 
        </FONT> BedRisers make it easy to get in and out of bed.<br><IMG src="images/bspace.gif"><li>Get a <FONT color=#ff0099><STRONG><u>better nights 
        sleep</u></STRONG></FONT>. (We don't know why.)<br><IMG src="images/bspace.gif"><li>Endless <FONT color=#ff0099> <STRONG><u>decorating</u></STRONG></FONT>  possibilities.<br><IMG src="images/bspace.gif"><li>Feel like a kid again sleeping in Grandma's 
        old, <FONT color=#ff0099>        
         <STRONG><u>high bed</u></STRONG></FONT>.
        <br><IMG src="images/bspace.gif"><LI><STRONG><FONT color=#ff0099><u>Very 
        Sturdy</u></FONT></STRONG>.&nbsp; Used in Nursing 
        Homes.<br></LI>          </ul></TD>
    <TD width="293">
      <p align=center><IMG alt="" border=0 height=220 src="images/bed0143.jpg" 
      width=293><br>
      <P align=center><IMG alt="" border=0 height=40 
      src="images/bedRiserlogo13.gif" width=164><br>
      <align=CENTER><EM><FONT size=1>A division of A-1 Machine &amp; Tool<br>Since 1959</FONT><br><FONT size=4><STRONG>(800) 
      679-1956</STRONG></FONT></EM></P>
      <P align=center><FONT size=4><STRONG><EM>24 Hours a 
      Day</EM></STRONG></FONT></P> </TD>
    <TD width="25%">
      <br><FONT 
      color=#0000ff><STRONG><i>As seen here:</i></STRONG></FONT><ul><li>QVC<li>Better Homes<li>Martha Stewart<li>Southern Living<li>Country Living<li>Woman's Day<li>Family Circle<li>McCall's
        <LI>Romantic Homes
        <LI>The Home Show
        <LI>Jan &amp; Nans Column
        <LI>Boston Globe
        <LI>Chicago Sun Times</LI></ul></TD></TR>
  <TR>
    <TD colSpan=3>
      <P align=center>&nbsp;</P>
      <P align=center>&nbsp;</P>
      <P align=center>&nbsp;</P>
      <P align=center><FONT size=5><STRONG>Turn Your Bed Into A Secret Storage 
      Compartment!</STRONG></FONT></P>
      <P align=center>&nbsp;</P></TD></TR>
  <TR>
    <TD><br><FONT 
      color=#0000ff><STRONG><i>Create Up To 63 Cubic Feet Of Extra Storage Under Your Bed!</i><br><br></STRONG></FONT><ul><li>Winter/Summer Clothes<li>Bicycle<li>Golf Bag<li>Arts &amp; Crafts<li>Extra Blankets &amp; Pillows<li>Christmas Decorations<li>Shoes<li>Pet Beds<li>Old Files</li> </ul></TD>
    <TD>
      <P align=center><IMG alt="" border=0 height=195 src="images/bed0393.jpg" 
      width=293></P></TD>
    <TD><br><FONT 
      color=#0000ff><STRONG><i>Ends Storage Problems For:</i><br></STRONG></FONT><ul><li>Apartments<li>Kids' Rooms<li>Condominiums<li>Mobile Homes<li>Small Homes<li>Older Homes with Small Closets<li>Cabins<li>College Dorm Rooms</li>  </ul></TD></TR>
  <TR>
    <TD colSpan=3>
      <P align=center><FONT size=4><STRONG>      
       </STRONG></FONT>&nbsp;</P>
      <P align=center><FONT size=4><STRONG></STRONG></FONT>&nbsp;</P>
      <P align=center><FONT size=4><STRONG></STRONG></FONT>&nbsp;</P>
      <P align=center><FONT size=5><STRONG>Transform Your Ordinary Bed Into An 
      Extraordinary Bed!!</STRONG></FONT></P>
      <P align=center>&nbsp;</P></TD></TR>
  <TR>
    <TD><br><ul><li>100% <FONT color=#ff0099><STRONG><u>Satisfaction 
        Guaranteed!</u></STRONG></FONT>  <li>Guaranteed to attach to your standard angle 
        iron bed frame or your <FONT color=#ff0099><STRONG><u>money 
        back!</u></STRONG></FONT>             <li><FONT color=#ff0099><STRONG><u>Installs in minutes</u></STRONG></FONT>   with only a cresent wrench!<li><FONT color=#ff0099><STRONG><u>No Drilling</u></STRONG></FONT>  Required!<li>Patented Design!<li>Made of <FONT color=#ff0099><STRONG><u>Industrial 
        Steel!</u> </STRONG></FONT>   </li>   </ul></TD>
    <TD>
      <P align=center><IMG alt="" border=0 height=218 
      src="images/bedRiserlegs.jpg" width=220></P></TD>
    <TD>
      <P align=center><FONT color=#ff0000><u><FONT 
      color=#ff0099>Important!!</FONT></u>
      </FONT> Check the<br>installation instructions<br>to be sure that you<br>have the right kind of<br>bed frame.<br><form action="http://www.bedriser.com/brinstall.htm" method="get"><INPUT name=Submit type=submit value="Go There!"></form></P></TD></TR>
  <TR>
    <TD colSpan=3>
      <DIV align=center>
      <TABLE><TR>
          <TD>
            <P align=center><STRONG>PLEASE READ</STRONG></P>
            <P>Dear Customer:</P>
            <P>Thank you for your recent interest in our BedRisers.&nbsp; A set 
            of BedRisers consists of six heavy duty steel legs and is available 
            in two different lengths.&nbsp; The set of six 12" or 15" legs will 
            fit any size standard, angle iron, metal bed frame.</P>
            <P>With the 12" BedRiser legs, your frame will be 12 inches from the 
            floor.&nbsp; Now measure your box springs and mattress, add 12 
            inches and that is how high your bed will be.&nbsp; The same goes 
            for the 15"&nbsp;BedRiser legs.&nbsp; Don't forget to take into 
            consideration how thick your quilt or bedspread is because if 
            they're very thick, that will make the bed look even 
        higher.</P></TD></TR></TABLE></DIV>
      <DIV align=center>&nbsp;</DIV></TD></TR>
  <TR>
    <TD colSpan=3>
      <DIV align=center>
      <TABLE width="60%"><TR>
          <TD><STRONG>IMPORTANT</STRONG>:&nbsp; The 15" BedRiser legs will 
            make your bed very high.&nbsp; Be sure to measure your box springs 
            and mattress then add 15", that's how high your bed will 
        be.</TD></TR></TABLE></DIV></TD></TR>
  <TR>
    <TD colSpan=3>
      <P>&nbsp;</P>


</font><FONT face=Arial>
      <P>
      <TABLE>
        <TBODY>
        <TR>
          <TD width="50%"></TD></FONT></FONT>
          <TD>
            <H2 align=center>Testimonials</H2></TD><FONT face=Arial><FONT face=Arial>
          <TD width="50%"></TD></TR>
        
        <TR>
          <TD width="50%">
            <HR align=center width="50%">
            <P><FONT size=2></FONT>            
                       
                  </P>
            <P><FONT size=2>Dear BedRiser,<br>It worked just fine - nice and 
            <FONT color=#ff0099><STRONG>       <u>strong and steady</u> </STRONG></FONT> - I looked for years for 
            such a product.&nbsp; Happy I found you in a magazine<EM>.</EM><br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
            Pamela</P></FONT>
            <HR align=center width="50%">

            <P></P><FONT size=2>
            <P><FONT size=2>Dear BedRiser,<BR>I love the BedRiser I 
            purchased.&nbsp; I am almost 6 feet tall with a&nbsp;<U><FONT color=#ff0099><STRONG>bad 
            back</STRONG></FONT> </U> and its so much easier for me to 
            get in and out of my bed now!&nbsp; And it looks great with my new 
            skirt.&nbsp; The BedRiser creates a romantic look with little 
            effort.&nbsp; Thank you for being so clever.<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Sherri</P>
            <P>     
                          
              </FONT></FONT><FONT size=2>
            <HR align=center width="50%">
            </FONT></TD>
          <TD><form action="http://www.bedriser.com/testimonials.htm" method="get"><INPUT name=test type=submit value="More Testimonials!"></form></TD>
          <TD width="50%">
            <HR align=center width="50%">

            <P><FONT size=2>            
                           
                         
              </P>
            <P><FONT size=2>Dear BedRiser,<BR>The BedRiser is wonderful.&nbsp; 
            Besides the <STRONG><FONT color=#ff0099>   
               <U>extra&nbsp;"closet"</U> </FONT></STRONG> our room 
            now looks special<EM>.</EM><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; A.J. 
            Mason</P>
            <P></FONT></FONT><FONT size=2>
            <HR align=center width="50%">
            </FONT>
            <P></P><FONT size=2>
            <P><FONT size=2>Dear BedRiser,<BR>Love my BedRiser, <STRONG><FONT 
            color=#ff0099>   <U>didn't 
            cost a lot</U> 
            </FONT></STRONG> of money to give our 
            bedroom a total new look.&nbsp; Easy, too!<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Jan</P>
            <P>     
                           
                         
                  </FONT></FONT><FONT size=2>
            <HR align=center width="50%">
            </FONT></TD></TR></TBODY></TABLE></P></FONT></FONT></TD></TR></TBODY></TABLE>
<P align=center><IMG alt="" border=0 height=250 src="images/bedskirt.jpg" 
width=293>&nbsp;
        <FORM action=https://www.bedriser.com/securedll/bedRiser.dll 
        method=post>
<TABLE width="100%" align='"CENTER"'>
  
  
  <TR>
    <TD>
      <P align=center><FONT color=#ff0099><STRONG>All plain dust ruffles are 
      very full and have a 200 thread count.</STRONG></FONT></P><FONT 
      color=#ff0099><STRONG>
      <DIV align=center>
      <TABLE bgColor=silver border=2>
        
        <TR>
          <TD><STRONG>
            <P align=center>
            <TABLE border=0 width=532>
              
              <TR>
                <TD><IMG alt="" border=0 height=30 src="images/sp99.gif" 
                  width=125></TD>
                <TD width="100%">
                  <P>12" BedRiser &amp; any color 18" bedskirt to fit&nbsp;any 
                  size bed.</P></TD></TR>
              <TR>
                <TD><IMG alt="" border=0 height=30 src="images/sp124.gif" 
                  width=125></TD>
                <TD>15" BedRiser &amp; any color 21" bedskirt to fit any size 
                  bed.</TD></TR></TABLE></P></STRONG>
            <DIV align=center>
            <TABLE border=1>
              
              <TR>
                <TD><STRONG>QTY</STRONG></TD>
                <TD>
                  <P align=center><STRONG><IMG alt="" border=0 height=5 
                  src="images/bspacegrey.gif" width=10>Select 
                  BedRiser</STRONG><IMG alt="" border=0 height=5 
                  src="images/bspacegrey.gif" width=10></P></TD>
                <TD>
                  <P align=center><STRONG>Select Bedskirt Color</STRONG></P></TD>
                <TD><STRONG>Select Size</STRONG></TD></TR>
              <TR>
                <TD>
                  <P align=center><INPUT maxLength=3 name=QTY size=2 
                value=1></P></TD>
                <TD><INPUT CHECKED name=bedRiser type=radio 
                  value=BR12><STRONG>12" BedRiser<BR></STRONG><FONT 
                  size=1>BR12NET</FONT><BR><INPUT name=bedRiser type=radio 
                  value=BR15><STRONG>15" BedRiser<BR></STRONG><FONT 
                  size=1>BR15NET</FONT></TD>
                <TD>
                  <TABLE>
                    
                    <TR>
                      <TD>
                        <P align=center><FONT size=1>BONE</FONT><BR><IMG alt="" 
                        border=0 height=45 src="images/beig.gif" 
                        width=45><BR><INPUT CHECKED name=color type=radio 
                        value=09></P></TD>
                      <TD>
                        <P align=center><FONT size=1>WHITE</FONT><BR><IMG alt="" 
                        border=0 height=45 src="images/whst.gif" 
                        width=45><BR><INPUT name=color type=radio 
                      value=97></P></TD>
                      <TD>
                        <P align=center><FONT size=1>NAVY</FONT><BR><IMG alt="" 
                        border=0 height=45 src="images/navfl.gif" 
                        width=45><BR><INPUT name=color type=radio 
                      value=65></P></TD>
                      <TD>
                        <P align=center><FONT size=1>BRGUNDY</FONT><BR><IMG 
                        alt="" border=0 height=45 src="images/Bgst.gif" 
                        width=45><BR><INPUT name=color type=radio 
                      value=11></P></TD>
                      <TD>
                        <P align=center><FONT size=1>ROSE</FONT><BR><IMG alt="" 
                        border=0 height=45 src="images/Mvst.gif" 
                        width=45><BR><INPUT name=color type=radio 
                      value=77></P></TD>
                      <TD>
                        <P align=center><FONT size=1>H GREEN</FONT><BR><IMG 
                        alt="" border=0 height=45 src="images/Emst.gif" 
                        width=45><BR><INPUT name=color type=radio 
                      value=39></P></TD></TR></TABLE></TD>
                <TD>
                  <P align=center><SELECT name=bedtype size=4><OPTION selected 
                    value=TWIN>Twin</OPTION> <OPTION value=FULL>Full</OPTION> 
                    <OPTION value=QUEEN>Queen</OPTION> <OPTION 
                    value=KING>King</OPTION></SELECT></P></TD></TR></TABLE></DIV>
            <P align=center>
            <TABLE border=0 style="HEIGHT: 46px; WIDTH: 532px" ?75%??>
              
              <TR>
                <TD>
                  <P align=center><FONT size=6><STRONG>Call NOW: (800) 
                  679-1956</STRONG></FONT></P>
                  <P align=center><FONT size=6><STRONG>24 Hours a Day / 7 Days a 
                  Week!</STRONG></FONT></P></TD></TR></TABLE></P></TD></TR></TABLE></DIV></STRONG></FONT></TD></TR></TABLE></P><form action="https://www.bedriser.com/securedll/bedRiser.dll" method="post"></FONT>
<P>
<TABLE bgcolor="silver">
  
  <TR>
    <TD>
      <DIV align=center>
      <TABLE border=1 width="100%">
        <TR>
          <TD></TD>
          <TD></TD></TR>
        <TR>
          <TD>
            <DIV align=left><STRONG>SHIPPING 
            INFORMATION&nbsp;</STRONG><FONT size=2>We ship via UPS.&nbsp; Parcel post is 
            used only if necessary.&nbsp; Sorry, no C.O.D's.&nbsp; Please call 
            <STRONG>(800) 679-1956 </STRONG>for orders to Hawaii, Alaska, Puerto 
            Rico, or Canada.&nbsp; Outside US orders please call <STRONG>(800) 
            679-1956 </STRONG>for ordering instructions and 
            prices.<STRONG>     
            &nbsp;</STRONG></FONT></DIV>
            <DIV align=left><FONT size=2><STRONG>$60.01 to 
            $100.00...............$8.99</STRONG></FONT></DIV>
            <DIV align=left><FONT size=2><STRONG>$100.01 to 
            $300.00...........$11.99</STRONG></FONT></DIV>
            <DIV align=left><FONT size=2><STRONG>$300.01+ Please call (800) 
            679-1956 to order.</STRONG></FONT></DIV>
            <DIV align=left><FONT size=2><FONT size=3><STRONG>ADDITIONAL 
            SHIPPING INFORMATION&nbsp; </STRONG></FONT>We make every effort to 
            get your order to you without delay, however, BedRisers are very 
            much in demand and sometimes we do have to backorder.&nbsp; We also 
            try to keep our bed ruffles and accessories in stock at all times, 
            but they are also on back order on occasion.&nbsp; We do not ship 
            partial orders unless you request it, because the full amount of 
            your purchase would have to be charged at the time of the first 
            shipment.</FONT></DIV>
            <DIV align=left><STRONG>PERSONAL CHECKS OR MONEY ORDERS&nbsp; 
            </STRONG><FONT size=2>Personal checks are allowed to clear two weeks 
            before we ship.&nbsp; Money orders are shipped right away if the 
            item is in stock.&nbsp; We DO NOT ship orders C.O.D.</FONT></DIV></TD>
          <TD>
            <DIV align=left><STRONG>HOW TO ORDER&nbsp; 
            </STRONG><FONT size=2>Submit this form or our <STRONG>Order 
            Department </STRONG>will take your order at <STRONG>(800) 
            679-1956</STRONG>, 24 hours a day, 7 days a week including 
            holidays.&nbsp; Our <STRONG>Customer Service Department 
            </STRONG>will be more than happy to answer any questions (no matter 
            how lengthy) or help solve any problems that may arise.&nbsp; We 
            will help you in selecting a bed ruffle and matching accessories or 
            mix and match too.&nbsp; Our customer service hours are Monday - 
            Friday, 8:00AM. to 4:30PM CST <STRONG>(800) 
            679-1956.</STRONG></FONT></DIV>
            <DIV align=left><STRONG>THE BEDRiser GUARANTEE</STRONG>&nbsp; <FONT 
            size=2>Everything you buy from BedRiser is 100% satisfaction guaranteed.&nbsp; If you are not 
            completely satisfied with a product, you may return it 
            for a courteous and prompt exchange or 
      refund.</FONT></DIV>
            <DIV align=left><STRONG>RETURNS&nbsp; </STRONG><FONT size=2>To 
            return items, complete the Return Form which was enclosed with your 
            order and mail back to us within 30 days.&nbsp; Your package must be 
            insured.&nbsp; Returned item(s) must be in good condition.&nbsp; If 
            you have lost your order form, just list your name, address, phone 
            number and the date of your purchase.&nbsp; It is the customer's 
            responsibility to pay shipping on returned merchandise.&nbsp; If the 
            wrong goods were sent or the goods you ordered arrived damaged, 
            please call us for a Return Merchandise Authorization for free 
            return 
shipping.</FONT></DIV></TD></TR></TABLE></DIV></TD></TR></TABLE></P>
<P>
</P></form>
</body>
</html>



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**************************************************************  
*       You are very welcome to participate in the           *
*                                                            *
*                      IEEE LANOMS99                         *
*                                                            *
* LATIN AMERICAN NETWORK OPERATIONS AND MANAGEMENT SYMPOSIUM *
*                                                            *
*            http://www.lrg.ufsc.br/~lanoms99/               *
*                                                            *
*                   Rio Atlantica Hotel                      *
*                Rio de Janeiro - Brazil                     *
*                   December 3-5, 1999                       *
**************************************************************

***** IEEE LANOMS99 - TECHNICAL SESSIONS ***** 

SESSION 1  - SECURITY MANAGEMENT 
Auditorium#1 - Dec 03 - 08:30 - 10:00 

1.1 - Access Control In Multicast Environments: An Approach 
to Senders Authentication 
      Antonio F. Gomez Skarmeta, Angel L. Mateo Martinez and Pedro M. 
      Ruiz Martinez 
      Universidad de Murcia (Spain) 

1.2 - A Large-scale System Authorization Scheme Proposal Integrating 
      Java, CORBA and Web Security Models and a Discretionary Prototype 
      Carla Merkle Westphall and Joni da Silva Fraga 
      Universidade Federal de Santa Catarina (Brazil) 

1.3 - Impact of Security Policies on the TMN-X Interface Integrity and 
      Performance. 
      Ognjen Prnjat and Lionel Sacks 
      University College London (England) 

SESSION 2 - MULTI-AGENTS AND MOBILE AGENTS 
Auditorium#2 - Dec 03 - 08:30 - 10:00 

2.1 - Multi-agent Advisory System. 
      Cuesta P., Barreiro E., Gonzalez E., Carrion P., 
      Laza R. and Corchado J. M. 
      Universidad de Vigo (Spain) 

2.2 - Traffic Management A Multi-Agent System Approach 
      Paulo Monteiro and Luis Correio 
      Alcatel Telecom (Portugal) 

2.3 - Analyzing Mobile Agent Scalability in Network Management 
      Marcelo Goncalves Rubistein and Otto Carlos Muniz Bandeira Duarte 
      Universidade Federal do Rio de Janeiro (Brazil) 

SESSION 3 - TELECOMMUNICATION POLICIES AND SOLUTIONS FOR NETWORK 
MANAGEMENT 
Auditorium#1 - Dec 03 - 10:30 - 12:00 
  
3.1 - Methodology for Specifying And Implementing a Management Solution 
      Sergio Clementi and Tereza Cristina Melo de Brito Carvalho 
      Universidade de São Paulo (Brazil) 
  
3.2 - Experiences in the Developing of a Prototype Sytem for the Exchange 
of Management Information Study Case: Colombia 
      Leonor Wilches-Chaux, Carlos E. Caciedo 
      CINTEL (Colombia) 

3.3 - The Foundations and Formation of Public Policies 
for Telecommunications 
      Keith von Mecklenburg 
      University of St. Andrews (Scothland) 

SESSION 4 - ARTIFICIAL INTELIGENCE SOLUTIONS FOR NETWORK MANAGEMENT 
Auditorium#2 - Dec 03 - 10:30 - 12:00 

4.1 - Solving the Parallel Task Scheduling Problem by Means of a 
Genetic Approach 
      Esquivel S.C., Gatica C.R. and Gallard R.H. 
      Universidad Nacional de San Luis (Argentina) 

4.2 - Enhanced Genetic Algorithms for the Cluster Allocation Problem in 
Distributed Systems 
      Apolloni R., Molina S. and Gallard R.H. 
      Universidad Nacional de San Luis (Argentina) 

4.3 - Fault Diagnosis for Local Area Network Environments 
      Volnys Bernal, Leliane N. de Barros, Marilza Lemos and 
      Jacques Wainer 
      Universidade de São Paulo / Universidade de Campinas (Brazil) 

SESSION 5 - ATM MANAGEMENT 
Auditorium#1 - Dec 03 - 14:30 - 15:30 

5.1 - Measure-based Estimation of The Required Capacity on ATM 
Switches Using Neural Networks 
      Miguel Franklin, Adriano Nascimento, Marcelino Pequeno 
      and Mauro Oliveira 
      Universidade Federal do Ceará / CEFET-CE (Brazil) 

5.2 - Comparative Analisys of Multicast Routing Algorithms 
for Multimedia Communication over ATM Networks 
      Frank Meylan, Luis Gustavo Gasparini Kiatake, Marcelo Zanoni 
Santos, 
      Sergio Takeo Kofuji and Jean Pierre Courtiart 
      LAAS-CNRS7 (France) / Universidade de São Paulo (Brazil) 

5.3 - Service Level Management in ATM Networks 
      Daniel Puka, Manoel Camillo Penna and Vinicius Prodocimo 
      CEFET-PR / Visionnaire (Brazil) 

SESSION 6 - TELECOMMUNICATION INFORMATION MANAGEMENT 
Auditorium#2 - Dec 03 - 14:30 - 15:30 

6.1 - Defining a Generic Information Model for Addresing New Service 
Offers in a Competitive Scenario 
     Jean-Daniel Guedj, Giovanni Martini and Luca Valtriani 
     Telecom Argentina (Argentina) / CSELT (Italy) 
  
6.2 - A process-driven methodological approach for the design of 
telecommunication management systems 
     Thierry Fraize, Julio Villena, Jean-Daniel Guedj 
     Telecom Argentina (Argentina) 

6.3 - Transmuting Legacy Network Management Environment 
into TMN-based INM 
      Wang-Dong Woo, Hee-Sam Hwang and Byung-Nam Yoon 
      Eletronics and Telecommunication Research Institute (Korea) 

SESSION 7 - TMN TECHNOLOGIES 
Auditorium#1 - Dec 04 - 08:30 - 10:00 

7.1 - Configuration Management Integration For Telecommunications 
Networks: An Implementation Experiment 
      Linnyer Beatrys Ruiz, Manoel Camillo Penna, Karlo Adriano Nocera, 
      Mario Jose Ferreira, Celso Antonio Alver Kaestner and Joelson dos 
Passos 
      PUC-PR (Brazil) 

7.2 - Experiential Knowledge in TMN 
      Alexandre Moraes Ramos and Joao Bosco da Mota Alves 
      Universidade do Vale do Itajaí / Universidade Federal de Santa 
Catarina (Brazil) 

7.3 - SMART-FO: TMN Maintenance Support System for Optical Fibre Network. 
      Francesco Russano 
      SIRTI (Italy) 

SESSION 8 - PROACTIVE AND PERFORMANCE MANAGEMENT 
Auditorium#2 - Dec 04 - 08:30 - 10:00 

8.1 - SEGRE: An Expert System for Pro-active Computer Network Management 
      Cecilia A. Castro Cesar and Celso de Renna e Souza 
      Instituto Tecnológico de Aeronáutica (Brazil) 

8.2 - Towards Proactive Network Load Management for Distributed Parallel 
Programming 
      Sergio Nesmachnow, Antonio Lopez and Carlos Lopez 
      Universidad de la República (Uruguay) 

8.3 - Performance Monitoring of Telemanagement Systems 
      Wagner Meira Jr, Patricia Aguiar, Jose Marcos Nogueira and 
      Christiano Mata Machado 
      Universidade Federal de Minas Gerais / Telemar Minas (Brazil) 

SESSION 9 - TRAFFIC MANAGEMENT 
Auditorium#1 - Dec 04 - 10:30 - 12:00 

9.1 - An Efficient Low-Cost Traffic Management System for Large Switching 
Networks 
      Ana Luiza Barros Diniz, Sergio Campos, Jose Marcos Nogueira, 
      Christiano Mata Machado and Luiz Geraldo Souza 
      Universidade Federal de Minas Gerais / Telemar Minas (Brazil) 

9.2 - Traffic Simulation and the Location of Mobile Units in Wireless 
Communication Systems 
      Mauro Nacif Rocha, Geraldo Robson Mateus and Soraia Lucia da Silva 
      Universidade Federal de Minas Gerais (Brazil) 

9.3 - Connection Admission Management for Self-Similar Traffic 
      Nelson L. S. Fonseca, Gilberto S. Mayor and Cesar A. V. Neto 
      Universidade de Campinas (Brazil) 

SESSION 10 - WEB-BASED MANAGEMENT AND INTERNET TECHNOLOGY 
Auditorium#2 - Dec 04 - 10:30 - 12:00 
  
10.1 - WebManager: A Web-Based Network Management Application 
       Jacques Philippe Sauve 
       Universidade Federal da Paraíba (Brazil) 

10.2 - Web Based Management in Lua 
       Edison Ishikawa, Noemi Rodriguez 
       IME / PUC-Rio (Brazil) 

10.3 - Reporting the Experience of a Post-Graduate Program in Computer 
Network Operations, Internet Technology and Operating Systems' Support 
       Manuel Lois Anido 
       Universidade Federal do Rio de Janeiro (Brazil) 

SESSION 11 - QUALITY OF SERVICE AND NETWORK PROVISION OPERATION 
Auditorium#1 - Dec 04 - 14:00 - 15:30 

11.1 - Resource Allocation and Quality of Service for Distributed 
Multimedia Applications 
       Carlos de Castro Goulart and Jose Marcos Silva Nogueira 
       Universidade Federal de Viçosa / Universidade Federal de Minas 
Gerais (Brazil) 

11.2 - Algorithms and Contracts for Network and Systems Management 
       Paulo Pereira and Paulo Pinto 
       Universidade Técnica de Lisboa / Universidade Nova de Lisboa 
(Portugal) 

11.3 - Work Force Management (WFM) 
       Bruno Guido, Gavazzi Roberto, Fabio Malabocchia and Luisella Sisto 
       CSELT (Italy) 

SESSION 12 - INTELLIGENT AGENT AND HIERARCHICAL MANAGEMENT SYSTEMS 
Auditorium#2 - Dec 04 - 14:00 - 15:30 

12.1 - An Environment for Developing Neural Network-based Intelligent 
Agents 
       Adriano Nascimento, Miguel Frankin and Mauro Oliveira 
       Universidade Federal do Ceará / CEFET-CE (Brazil) 

12.2 - Distributed Network Security Management Using Intelligent Agents 
       K. Boudaoud and N. Agoulmine 
       EURECOM Institute / PRISM Laboratory (France) 

12.3 - Hierarchical Management Protocol for Large Networks 
       Marcial Porto Fernandez, Gardel Moreira Delfino and 
       Aloysio de Castro P. Pedroza 
       Universidade Federal do Rio de Janeiro (Brazil) 
  


***** IEEE LANOMS99 - PANELS *****

PANEL1  - TELECOMMUNICATIONS OPERATIONS AND MANAGEMENT IN LATIN AMERICA 
Auditorium#1 - Dec 03 - 16:00 - 18:00 

PANEL2  - MOBILE TELECOMMUNICATIONS: TRENDS AND OPPORTUNITIES 
Auditorium#1 - Dec 04 - 16:00 - 18:00 


***** IEEE LANOMS99 - TUTORIALS *****

TUTORIAL 1.1 - The technological and Economic Effects of the Deregulation
               of the Global Telecommunications Industry upon the Application
               of the Principles of Network Management
Auditorium#1 - Dec 05 - 09:00 - 12:30

TUTORIAL 1.2 - Neural Networks or Distributed Algorithms for Network
               Management
Auditorium#2 - Dec 05 - 09:00 - 12:30 

TUTORIAL 2.1 - New Global Solutions of Event Correlation based on 
               Distributed  infrastructure
Auditorium#1 - Dec 05 - 14:00 - 17:30 

TUTORIAL 2.2 - Network Management with SNMP: The Manager-Agent and
               Distributed Paradigms
Auditorium#2 - Dec 05 - 14:00 - 17:30 


***** IEEE LANOMS99 - REGISTRATION FORM ****

NAME:
TITLE:
ORGANIZATION:
DEPARTMENT:
ADDRESS:
STATE:
COUNTRY:
TELEPHONE:
FAX:

Full or Technical Sessions IEEE Members Registration - Member Number:
Full (Technical Sessions + 2 Tutorials)
( ) USD 499 on / before Nov. 26  or ( ) USD 559 after Nov. 26
Technical Sessions
( ) USD 290 Profissional or ( ) USD 190 Authors or ( ) USD 60 Students

Full or Technical Sessions IEEE NON-Members Registration
Full (Technical Sessions + 2 Tutorials)
( ) USD 499 on / before Nov. 26 or ( ) USD 559 after Nov. 26
Technical Sessions
( ) USD 350 Profissional or ( ) USD 190 Authors or ( ) USD 70 Students

Number of tutorial to be Registrated
IEEE Members     ( ) USD 125 - 1 Tutorial or ( ) USD 225 - 2 Tutorials
IEEE Non-Members ( ) USD 150 - 1 Tutorial or ( ) USD 250 - 2 Tutorials
Tutorials (Mark your choices)
( ) Tutorial 1.1 or ( ) Tutorial 1.2
( ) Tutorial 2.1 or ( ) Tutorial 2.1

Proceedings (IEEE Members / Non-members)
( ) USD 30 

Total Registration: USD______
Total Tutorial: USD________
Total Proceedings: USD_______
Total Reminttance : USD_______

Method of Payment
( ) Method 1 - Money order in USD Payable to FEESC
( ) Method 2 - Check Brazilian Banks in R$ (exchange rate 1,9 R$/USD)
For both methods make transfer to the following bank account:
Bank Name: Banco do Brasil
Bank Number: 001
Branch Number: 1453-2
Bank Account: 204.422.6
( ) Method 3 - ( ) Visa or ( ) Mastercard
Number:
Expiration Date:
Account Name:
Authorized ( ) I authorize FEESC to deduct the amount inserted at
TOTAL REMITTANCE from my credit card.
 
SEND BY FAX OR BY EMAIL THE COMPLETED FORM WITH PAYMENT TO:
Fundacao de Ensino e Engenharia de Santa Catarina (FEESC)
Caixa Postal 5040
88040-970, Florianopolis (SC) Brazil
Phone: +55.48.2334455
FAX: +55.48.3319677
EMAIL: katia@feesc.ufsc.br


***** IEEE LANOMS99 - COMMITTEE MEMBERS *****

General and  TPC Chair: 
Carlos Becker Westphall, Brazil 

TPC Co-Chairs: 
Luiz Fernando Kormann, Brazil 
Mirela Sechi Moretti Annoni Notare, Brazil 

Publication Chair: 
Bernardo Goncalves Riso, Brazil 

Publicity Chair: 
Juan Carlos Miguez, Uruguay 

Finance Chair: 
Fernando Augusto da Silva Cruz, Brazil 
Joao Bosco Mangueira Sobral, Brazil 

Local Arrangements Chair: 
Michael Stanton, Brazil 

IEEE CNOM Chair: 
Veli Sahin, USA 

International Liasions: 
Africa: J. Roos, South Africa 
Asia: J. Hong, Korea 
Europe: E. Bagnasco, Italy 
N. America: S. Aidarous, USA 
Oceania: P. Ray, Australia 

Advisory Board: 
B. Meandzijas, USA 
D. Zuckerman, USA 
M. Ejiri, Japan 
M. Yoshida, Japan 
R. Saracco, Italy 
S. Goyal, USA 
V. Sahin, USA 
W. Zimmer, Germany 

Technical Program Committee 
Adarsh Sethi (USA) 
Alan Mckinlay (UK) 
Alejandro Miralles (Chile) 
Behcet Sarikaya (Japan) 
Carlos de Lara (Mexico) 
Carlos E. Caicedo (Colombia) 
Edward Pinnes (USA) 
Edmundo R. Madeira (Brazil) 
Elias Procopio Duarte Junior (Brazil) 
Gabriel Jakobson (USA) 
German Goldszmidt (USA) 
Guy Pujolle (France) 
Heinz-Gerd Hegering (Germany) 
Joberto S. Barbosa Martins (Brazil) 
Jong-Tae Park (Korea) 
Jose Marcos Nogueira (Brazil) 
Jose Neuman de Souza (Brazil) 
Keith-Stuart von Mecklenburg (UK) 
Leonor W. Chaux (Colombia) 
Manu Malek (USA) 
Manoel C. Penna (Brazil) 
Michael Bauer (Canada) 
Michelle Sibilla (France) 
Nelson Fonseca (Brazil) 
Osvaldo Bianchi (Argentina) 
Paulo Gomide Cohn (Brazil) 
Raul Monge (Chile) 
Robert Weihmayer (USA) 
Roberto Vercelli (Italy) 
Seraphin Calo (USA) 
Varoozh Harikian (Belgium) 


























From mohwo@att.net  Sun Oct 24 04:16:39 1999
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Here is what you will receive when you get your first Package.  
Inside you will find 100 envelopes, 100 labels and 100 sales 
letters ready to stuff and mail

As soon as you are done with stuffing and mailing these first 
letters, your payment will arrive shortly, thereafter. All you 
have to do is to order more free supplies and stuff and mail more 
envelopes to make more money.

Our sales literature which you will be stuffing and mailing will 
contain
information outlining our highly informative manuals that we are
advertising nationwide.  As a free gift you will receive a 
special manual valued at  $24.95, absolutely free, just for 
joining our Home Mailers Program.

Plus you will get your own special code number, so that we will 
know how much you are to get paid.  And to make re-ordering of 
more envelopes, that our company supplies very simple for you.

We are giving you this free bonus because we want you to be 
confident in our company and to ensure that we will be doing 
business with you for a long time.

Benefits Of This Job:

1. You do not have to quit your present job, to earn more money 
at home
2. You can make between $2,500 to $4,500 a month depending on the 
amount of time you are willing to spend stuffing and mailing 
envelopes
3. This is a great opportunity for the students, mothers, 
disabled persons or those who are home bodies.

To secure your position and to show us that you are serious about 
earning extra income at home we require a one-time registration 
fee of $35.00.
This fee covers the cost of your initial start up package,  which 
includes 100 envelopes, 100 labels and 100 sales letters and a 
manual, your registration fee will be refunded back to you 
shortly thereafter.

Money Back Guarantee!

We guarantee that as soon as you stuff and mail your first 300 
envelopes You will be paid $75.00 and your registration fee will 
be refunded.

Many of you wonder why it is necessary to pay a deposit to get a 
job. It is because we are looking for people that seriously want 
to work from home.  

*  If 3.000 people told us they wanted to start working from home 
and we sent out 3.000 packages free to every one.  And then half 
of the people decided not to work, this would be a potential loss 
of more than $60,000 in supply's and shipping that we have sent 
out to people that don't want to work

We have instituted this policy to make sure that you really want 
to work and at least finish your first package.

To Get Started Today Please Enclose Your Registration Fee of $35
Check, Or Money Order and fill out the application below and 
mail to:

MOHW Co	
PMB
11054 Ventura Blvd #126	
Studio City, CA 91604

Name_____________________________________________________

Address___________________________________________________

City____________________________________ State______________

Zip Code________________

Telephone Number(s)_________________________________________

E-mail Address______________________________________________



For all orders, please allow seven (7) days for delivery and up 
to 10 days. Money Orders will result in faster shipping of your package.
 
 
 
 
 
 
 
 


From rem-conf-request@es.net  Sun Oct 24 04:26:38 1999
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From: <mohwo@att.net>
Subject: We need Home Workers!
Date: Sun, 24 Oct 1999 02:44:31
Message-Id: <663.820814.160372@ns.bigbear.com>
To: rem-conf@es.net
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Dear Future Associate,

You Can Work At Home & Set Your Own Hours.  Start earning Big 
Money in a short time
       
                                    NO Newspaper Advertising!

Your job will be to stuff and mail envelopes for our company. You 
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out.

Just follow our simple instructions and you will be making money 
as easy as
1… 2… 3

For example stuff and mail 200 envelopes and you will receive 
$50.00. Stuff and mail 1000 and you will receive $250.00. Stuff 
and mail 2000 and you will receive $500.00 and more 

Never before has there been an easier way to make money from 
home!

Our Company's Home Mailing Program is designed for people with 
little or no experience and provides simple, step by step 
instructions.  

There is no prior experience or special skills necessary on your 
part, Just stuffing envelopes.

We need the help of honest and reliable home workers like you.  
Because we are overloaded with work and have more than our staff 
can handle. We have now expanded our mailing program and are 
expecting to reach millions more with our offers throughout the 
US and Canada.

Our system of stuffing and mailing envelopes is very simple and 
easy to do!
You will not be required to buy envelopes or postage stamps.

We will gladly furnish all circulars at no cost to you. We assure 
you that as a participant in our program you will never have to 
mail anything objective or offensive. 

There are no quotas to meet, and there no contracts to sign. You 
can work as much, or as little as you want. Payment for each 
envelope you send out is Guaranteed!

Here is what you will receive when you get your first Package.  
Inside you will find 100 envelopes, 100 labels and 100 sales 
letters ready to stuff and mail

As soon as you are done with stuffing and mailing these first 
letters, your payment will arrive shortly, thereafter. All you 
have to do is to order more free supplies and stuff and mail more 
envelopes to make more money.

Our sales literature which you will be stuffing and mailing will 
contain
information outlining our highly informative manuals that we are
advertising nationwide.  As a free gift you will receive a 
special manual valued at  $24.95, absolutely free, just for 
joining our Home Mailers Program.

Plus you will get your own special code number, so that we will 
know how much you are to get paid.  And to make re-ordering of 
more envelopes, that our company supplies very simple for you.

We are giving you this free bonus because we want you to be 
confident in our company and to ensure that we will be doing 
business with you for a long time.

Benefits Of This Job:

1. You do not have to quit your present job, to earn more money 
at home
2. You can make between $2,500 to $4,500 a month depending on the 
amount of time you are willing to spend stuffing and mailing 
envelopes
3. This is a great opportunity for the students, mothers, 
disabled persons or those who are home bodies.

To secure your position and to show us that you are serious about 
earning extra income at home we require a one-time registration 
fee of $35.00.
This fee covers the cost of your initial start up package,  which 
includes 100 envelopes, 100 labels and 100 sales letters and a 
manual, your registration fee will be refunded back to you 
shortly thereafter.

Money Back Guarantee!

We guarantee that as soon as you stuff and mail your first 300 
envelopes You will be paid $75.00 and your registration fee will 
be refunded.

Many of you wonder why it is necessary to pay a deposit to get a 
job. It is because we are looking for people that seriously want 
to work from home.  

*  If 3.000 people told us they wanted to start working from home 
and we sent out 3.000 packages free to every one.  And then half 
of the people decided not to work, this would be a potential loss 
of more than $60,000 in supply's and shipping that we have sent 
out to people that don't want to work

We have instituted this policy to make sure that you really want 
to work and at least finish your first package.

To Get Started Today Please Enclose Your Registration Fee of $35
Check, Or Money Order and fill out the application below and 
mail to:

MOHW Co	
PMB
11054 Ventura Blvd #126	
Studio City, CA 91604

Name_____________________________________________________

Address___________________________________________________

City____________________________________ State______________

Zip Code________________

Telephone Number(s)_________________________________________

E-mail Address______________________________________________



For all orders, please allow seven (7) days for delivery and up 
to 10 days. Money Orders will result in faster shipping of your package.
 
 
 
 
 
 
 
 
 



From rem-conf-request@es.net  Sun Oct 24 20:34:17 1999
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From: "Jon" <rqpq@newmail.net>
Subject: And What?
To: misc93i@router.ti.itb.ac.id
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From rem-conf-request@es.net  Sun Oct 24 20:35:19 1999
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Message-ID: <008a01bf1e7d$2d329c20$4b51c485@ef63.eel.rdc.toshiba.co.jp>
From: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>
To: "Yoshihiro Kikuchi" <kiku@eel.rdc.toshiba.co.jp>, <rem-conf@es.net>,
        "4on2andIP-sys" <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: Submission of I/D: RTP payload format for MPEG-4 Audio/Visual streams
Date: Mon, 25 Oct 1999 09:09:15 +0900
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Dear all,

I'm sorry, I forgot to attach the document.

Regards,
Yoshihiro

-----Original Message-----
From: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
To: rem-conf@es.net <rem-conf@es.net>; 4on2andIP-sys
<4on2andIP-sys@advent.ee.columbia.edu>
Cc: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
Date: Saturday, October 23, 1999 12:34 PM
Subject: Submission of I/D: RTP payload format for MPEG-4 Audio/Visual
streams


>Dear all,
>
>I submitted an Internet draft for the RTP payload format for MPEG-4
>Audio/Visual streams:
>
>Title : RTP payload format for MPEG-4 Audio/Visual streams
>Author(s) : Yoshihiro Kikuchi, Toshiyuki Nomura, Shigeru Fukunaga,
>                 Yoshinori Matsui, Hideaki Kimata
>Filename : draft-jnb-mpeg4av-rtp-00.txt
>Pages : 14
>Date : 22-Oct-99
>
>This document describes RTP payload formats for the carriage of MPEG-4
>Audio and Visual streams, and an RTCP format for MPEG-4 backward
channel
>messages functionalities. The specification is based on the study on
>Internet draft for the carriage of MPEG-4 on IP, which is sent from
>ISO/IEC JTC1/SC29/WG11(MPEG) as a liaison statement. In this
>specification, MPEG-4 Audio/Visual bitstreams are directly mapped into
>RTP packets without using MPEG-4 Systems. The RTP header fields usage
>and the fragmentation rule for MPEG-4 Visual and Audio bitstreams are
>specified. It also specifies an RTCP packet usage to carry the MPEG-4
>backward channel messages.
>
>
>
>I attach this document. Review and comments by the experts are much
>appreciated.
>
>Best regards,
>Yoshihiro Kikuchi
>
>----
>        Yoshihiro Kikuchi
>
>E-mail: kiku@eel.rdc.toshiba.co.jp
>Corporate Research and Development Center
>TOSHIBA Corporation
>TEL: +81 +44 549 2288    FAX: +81 +44 520 1308
>
>
>

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Internet Engineering Task Force             Yoshihiro Kikuchi - Toshiba
Internet Draft                                   Toshiyuki Nomura _ NEC
Document: draft-jnb-mpeg4av-rtp-00.txt           Shigeru Fukunaga _ Oki
                                          Yoshinori Matsui _ Matsushita
                                                   Hideaki Kimata _ NTT
                                                       October 22, 1999


             RTP payload format for MPEG-4 Audio/Visual streams


Status of this Memo

   This document is an Internet-Draft and is in full conformance with =
all
      provisions of Section 10 of RFC2026 [1].

   Internet-Drafts are working documents of the Internet Engineering =
Task
   Force (IETF), its areas, and its working groups. Note that other =
groups
   may also distribute working documents as Internet-Drafts. =
Internet-Drafts
   are draft documents valid for a maximum of six months and may be =
updated,
   replaced, or obsoleted by other documents at any time. It is
   inappropriate to use Internet- Drafts as reference material or to =
cite
   them other than as "work in progress."
   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt
   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.






                                   Abstract

   This document describes RTP payload formats for the carriage of =
MPEG-4
   Audio and Visual streams[2][3], and an RTCP format for MPEG-4 =
backward
   channel messages functionalities[4]. It is based on the RTP/RTCP
   specification descried in the study on Internet draft for the =
carriage of
   MPEG-4 on IP[5]. In this specification, MPEG-4 Audio/Visual =
bitstreams
   are directly mapped into RTP packets without using MPEG-4 Systems[6]. =
The
   RTP header fields usage and the fragmentation rule for MPEG-4 Visual =
and
   Audio bitstreams are specified. It also specifies an RTCP packet =
usage to
   carry the MPEG-4 backward channel messages.




1. Introduction

1.1 Why MPEG-4 Audio/Visual RTP format needed?

   The RTP payload formats described in this Internet-Draft specify the
   normative way on how MPEG-4 Audio/Visual streams are fragmented and
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D


   mapped directly onto RTP packets. No extra header field is used for =
such
   functionality as error protection or grouping of streams.

   H.323 terminals could be the case.  MPEG-4 Audio/Visual streams are =
not
   managed by Object Descriptors[6] but H.245, and directly mapped into =
RTP
   packets without Sync Layer[6]. The semantics of RTP headers in this =
case
   need to be clearly defined including the association with the MPEG-4
   Audio/Visual data elements.  In addition, it would be beneficial to
   define the fragmentation rule of RTP packets for MPEG-4 Video streams =
to
   enhance error resiliency by utilizing the error resilience tools =
provided
   inside the MPEG-4 Video stream.  However, they have not been studied
   until now.


1.2 Consideration on the MPEG-4 Visual RTP payload format

   Considering that MPEG-4 Visual is used on a wide variety of networks =
from
   several Kbps to many Mbps, from guarantied networks with almost =
error-
   free to mobile networks with high error rate, it is not desired to =
apply
   too much restriction on the fragmentation like a single video packet
   shall always be mapped on a single RTP packet. On the other hand, a
   careless media unaware fragmentation will cause degradation of the =
error
   resiliency and the bandwidth efficiency. The fragmentation rule =
described
   in this document is flexible but to define the minimum rules to =
prevent
   the meaningless fragmentation of e.g. splitting a header into =
packets.

   For video coding media such as H.261 or MPEG-1/2, the additional =
media
   specific RTP header works effectively for recovering e.g. a picture
   header corrupt by packet losses. However, there is an error =
resilience
   functionality inside MPEG-4 Visual to recover corrupt headers. This
   functionality can commonly be used on RTP/IP network as well as other
   networks. (H.223/mobile, MPEG-2/TS, etc.) Therefore, there is no =
strong
   reason to define MPEG-4 Visual specific extra RTP header fields.


1.3 Consideration on the MPEG-4 Audio RTP payload format

   MPEG-4 Audio is a new kind of audio standard that integrates many
   different types of audio coding tools. It also supports a mechanism
   representing synthesized sounds. Low-overhead MPEG-4 Audio Transport
   Multiplex (LATM) manages the sequence of the compressed or the
   represented audio data by MPEG-4 Audio tools with relatively small
   overhead. In audio-only applications, the LATM-based MPEG-4 Audio
   bitstreams, therefore, are desirable to be directly mapped into the =
RTP
   packets without using MPEG-4 Systems.

   Furthermore, if the payload of a packet is a single audio frame, a =
packet
   loss does not impair the decodability of adjacent packets. Therefore, =
a
   payload specific header for MPEG-4 Audio is not required as same as =
one
   for the other audio coders.
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D


1.4 MPEG-4 Audio/Visual backward channel messaging on RTCP packets

   In some cases, MPEG-4 Audio/Visual has backward channel messaging
   functionalities. These messages are extremely Audio/Visual specific,
   since coders directly use these messages for controlling coding
   parameters. From the point of view of controlling parameters, these
   messages should be transmitted without delay. Therefore these =
messages
   are directly mapped onto some kind of low delay RTCP packets.



2. Conventions used in this document

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC-2119 [7].



3. RTP Packetization of MPEG-4 Visual bitstream

   This section specifies the RTP packetization rule for MPEG-4 Visual
   content. An MPEG-4 Visual bitstream is mapped directly onto the RTP
   payload without any addition of extra header fields or removal of any
   Visual syntax elements. The Combined Configuration/Elementary streams
   mode is used so that the configuration information is carried in the =
same
   RTP port as the elementary stream. (see 6.2.1 "Start codes" of =
ISO/IEC
   14496-2 [2])

   When the short video header mode is used, RTP payload format for =
H.263
   specified in the relevant RFCs or other standards MAY be used.


   0                   1                   2                   3
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=3D2|P|X|  CC   |M|     PT      |       sequence number         | =
RTP
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           timestamp                           | =
Header
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |            contributing source (CSRC) identifiers             |
   |                             ....                              |
   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |                                                               | RTP
   |       MPEG-4 Visual stream (byte aligned)                     | =
Payload
   |                                                               |
   |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                               :...OPTIONAL RTP padding        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

        Figure 1 - An RTP packet for MPEG-4 Visual stream
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D




3.1 RTP header fields usage for MPEG-4 Visual

   Payload Type (PT): Distinct payload type should be assigned to =
specify
   MPEG-4 Visual RTP payload format. If the dynamic payload type =
assignment
   is used, it is specified by some out-of-band means (e.g. H.245, SDP,
   etc.) that the MPEG-4 Visual payload format is used for the =
corresponding
   RTP packet.


   Extension (X) bit: Defined by the RTP profile used.


   Sequence Number: Increment by one for each RTP data packet sent. It
   starts with a random initial value for security reasons.


   Marker (M) bit: The marker bit is set to one when the RTP packet =
contains
   the configuration information (see 6.2.1 "Start codes" of ISO/IEC =
14496-2
   [2]), Group_of_VideoObjectPlane(), and/or the first fragment of an =
intra-
   coded VOP. Otherwise, set to zero.



   This bit is used to indicate that the RTP packet is a point for =
random
   access, error-recovery using an intra-coded VOP, and recovering the
   configuration information.


   Timestamp: The timestamp indicates the composition time, or the
   presentation time in a no-compositor decoder by adding a constant =
random
   offset for security reasons. For a video object plane, it is defined =
by
   vop_time_increment (in units of 1/vop_time_increment_resolution =
seconds)
   plus the cumulative number of whole seconds specified by =
module_time_base
   and time_code of Group_of_VideoObjectPlane() if present. In the case =
of
   interlaced video, a VOP consists of lines from two fields and the
   timestamp indicates the composition time of the first field. If the =
RTP
   packet contains only configuration information and/or
   Group_of_VideoObjectPlane(), the composition time of the subsequent =
VOP
   in the coding order is used.

   Unless specified by an out-of-band means, the resolution of the =
timestamp
   is set to its default.


   SSRC, CC and CSRC fields are used as described in RFC 1889 [8].


3.2 Fragmentation of MPEG-4 Visual bitstream

   A fragmented MPEG-4 Visual bitstream is mapped directly onto the RTP
   payload without any addition of extra header fields or removal of any
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D


   Visual syntax elements. The Combined Configuration/Elementary streams
   mode is used. The following rules apply for the fragmentation.

   (1) A header of the configuration information function and
   Group_of_VideoObjectPlane() SHALL be placed at the beginning of the =
RTP
   payload (just after the RTP header) or just after the header of the
   syntactically upper layer function.

   (2) If one or more headers exist in the RTP payload, the RTP payload
   SHALL begin with the header of the syntactically highest function.

   (3) A header SHALL NOT be split into a plurality of RTP packets.

   (4) Two or more VOPs SHALL be fragmented into different RTP packets =
so
   that one RTP packet consists of the data bytes associated with an =
unique
   presentation time (that indicated to the timestamp field in the RTP
   packet header).

   (5) A single video packet SHOULD NOT be split into a plurality of RTP
   packets. The size of a video packet SHOULD be adjusted such that the
   resulting RTP packet is not larger than the path-MTU.


   Here, header means the header of the configuration information =
functions
   (VisualObjectSequence(), VisualObject(), VisualObjectLayer()) and the
   entry point functions for a elementary stream
   (Group_of_VideoObjectPlane(), VideoObjectPlane(),
   video_plane_with_short_header(), MeshObject(), FaceObject()), the =
header
   of gob_layer(), and video_packet_header(). (see 6.2.1 "Start codes" =
of
   ISO/IEC 14496-2 [2])


3.3 Examples of packetized MPEG-4 Visual bitstream

   Considering that MPEG-4 Visual is used on a wide variety of networks =
from
   several Kbps to many Mbps, from guarantied networks with almost =
error-
   free to mobile networks with high error rate, it is not desired to =
apply
   too much restriction on the fragmentation like a single video packet
   shall always be mapped on a single RTP packet. On the other hand, a
   careless media unaware fragmentation will cause degradation of the =
error
   resiliency and the bandwidth efficiency. The fragmentation criteria
   described in 3.2 are flexible but to define the minimum rules to =
prevent
   the meaningless fragmentation of e.g. splitting a header into =
packets.

   For video coding media such as H.261 or MPEG-1/2, the additional =
media
   specific RTP header works effectively for recovering e.g. a picture
   header corrupt by packet losses. However, there is an error =
resilience
   functionality inside MPEG-4 Visual to recover corrupt headers. This
   functionality can commonly be used on RTP/IP network as well as other
   networks. (H.223/mobile, MPEG-2/TS, etc.) Therefore, there is no =
strong
   reason to define MPEG-4 Visual specific extra RTP header fields.
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D


   Figure 2 shows examples of RTP packets generated based on the =
criteria
   described in 3.2

   (a) is an example of the first RTP packet of an MPEG-4 visual object
   sequence. This packet contains the configuration information. =
According
   to the criterion (1), the header of VisualObjectSequence(VS header) =
is
   placed at the beginning of the RTP payload, and the headers of
   VisualObject and VisualObjectLayer(VO header, VOL header) follow it.

   (b) is an example the RTP packet that contains
   Group_of_VideoObjectPlane(GOV). Following the criterion (1), the GOV =
is
   placed at the beginning of the RTP payload. It is a waste of =
RTP/UDP/IP
   header overhead to generate a RTP packet containing only a GOV whose
   length is 7 bytes. Therefore, (a part of) the following VOP can be =
placed
   in the same RTP packet as shown in (b).

   (c) is an example that one video packet is packetized into one RTP
   packet. When the packet-loss rate of the underlying network is high, =
this
   kind of packetization is recommended. It is strongly recommended to =
set
   resync_marker_disable to 0 in the VOL header to enable adjustment of =
the
   video packet size. Even when the RTP packet containing the VOP header =
is
   discarded by a packet loss, the other RTP packets can be decoded by =
using
   the HEC(Header Extension Code) information in the video packet =
header. No
   extra RTP header field is necessary.

   (d) is an example that more than one video packets are packetized =
into
   one RTP packet. This kind of packetization is effective to save the
   overhead of RTP/UDP/IP headers if the bit-rate of the underlying =
network
   is low. However, it will decrease the packet-loss resiliency because
   multiple video packets are discarded by a single RTP packet loss. The
   adequate number of video packets in a RTP packet and the RTP packet
   length depend the packet-loss rate and the bit-rate of the underlying
   network.

   Figure 3 shows examples of RTP packets prohibited by the criteria of =
3.2.

   Fragmentation of a header into multiple RTP packets, like (a), will =
not
   only increase the overhead of RTP/UDP/IP headers but also decrease =
the
   error resiliency. Therefore, it is prohibited by the criterion (3).

   When concatenating more than one video packets into a RTP packet, VOP
   header or video_packet_header() shall not be placed in the middle of =
the
   RTP payload. The packetization like (b) is not allowed by the =
criterion
   (2). This is because of the error resiliency. Comparing this example =
with
   Figure 2(c), two video packets are mapped onto two RTP packets in =
both
   cases. However, there is a difference between the packet-loss =
resiliency.
   When the second RTP packet is lost, both video packets 1 and 2 are =
lost
   in the case of Figure 3(b) whereas only video packet 2 is lost in the
   case of Figure 2(c).

   A RTP packet containing more than one VOPs, like (c), is not allowed.
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D


       +------+------+------+------+
   (a) | RTP  |  VS  |  VO  | VOL  |
       |header|header|header|header|
       +------+------+------+------+

       +------+-----+------------------+
   (b) | RTP  | GOV |Video Object Plane|
       |header|     |                  |
       +------+-----+------------------+

       +------+------+------------+  +------+------+------------+
   (c) | RTP  | VOP  |Video Packet|  | RTP  |  VP  |Video Packet|
       |header|header|    (1)     |  |header|header|    (2)     |
       +------+------+------------+  +------+------+------------+

       =
+------+------+------------+------+------------+------+------------+
   (d) | RTP  |  VP  |Video Packet|  VP  |Video Packet|  VP  |Video =
Packet|
       |header|header|     (1)    |header|    (2)     |header|    (3)    =
 |
       =
+------+------+------------+------+------------+------+------------+

        Figure 2 - Examples of RTP packetized MPEG-4 Visual bitstream


       +------+-------------+  +------+------------+------------+
   (a) | RTP  |First half of|  | RTP  |Last half of|Video Packet|
       |header|  VP header  |  |header|  VP header |            |
       +------+-------------+  +------+------------+------------+

       +------+------+----------+  =
+------+---------+------+------------+
   (b) | RTP  | VOP  |First half|  | RTP  |Last half|  VP  |Video =
Packet|
       |header|header| of VP(1) |  |header| of VP(1)|header|    (2)     =
|
       +------+------+----------+  =
+------+---------+------+------------+

       +------+------+------------------+------+------------------+
   (c) | RTP  | VOP  |Video Object Plane| VOP  |Video Object Plane|
       |header|header|        (1)       |header|       (2)        |
       +------+------+------------------+------+------------------+

   Figure 3 - Examples of prohibited RTP packetization for MPEG-4 Visual
   bitstream




4. RTP Packetization of MPEG-4 Audio bitstream

   When tools defined in MPEG-4 Systems are not used MPEG-4 Audio stream =
is
   formatted by LATM (Low-overhead MPEG-4 Audio Transport Multiplex)
   format[9], and then mapped onto RTP packets as described the =
subsequent
   section.

4.1 RTP Packet Format
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D


   The LATM consists of the sequence of audioMuxElements that include =
one or
   more audio frames. A complete audioMuxElement or the part of
   audioMuxElements SHALL be mapped directly onto the RTP payload =
without
   removal of any audioMuxElement syntax elements as shown in Figure 4. =
The
   first byte of each audioMuxElement SHALL be located at the first =
payload
   location of an RTP packet.


   0                   1                   2                   3
   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=3D2|P|X|  CC   |M|     PT      |       sequence number         =
|RTP
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           timestamp                           =
|Header
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           synchronization source (SSRC) identifier            |
   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |            contributing source (CSRC) identifiers             |
   |                             ....                              |
   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
   |                                                               |RTP
   :                 audioMuxElement (byte aligned)                =
:Payload
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                Figure 4 - An RTP packet for MPEG-4 Audio

   It is required for the audioMuxElement to indicate the following
   information by an out-of-band means.

   dataTypePresentEnable: If this information is ON, audioMuxElement =
SHALL
   include an indication bit "dataType". This bit indicates the data =
type of
   audioMuxElement is whether "setupData" element or "payloadMux" =
element.
   If this information is OFF, audioMuxElement contains the payloadMux
   element.

   errorProtectionEnable: This information indicates whether error
   protection is used.

   Furthermore, the length of the audioMuxElement in bytes SHALL be =
required
   for the LATM-based audio decoder. This length MAY be equal with the
   length of the RTP payload. If the audioMuxElement is fragmented into
   several RTP packets, the cumulative length of all RTP packets MAY be
   used.

4.2 RTP Header Fields Usage

   Payload Type (PT): Distinct payload type should be assigned to =
specify
   MPEG-4 Audio RTP payload format. If the dynamic payload type =
assignment
   is used, it is specified by some out-of-band means (e.g. H.245, SDP,
   etc.) that the MPEG-4 Audio payload format is used for the =
corresponding
   RTP packet.
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D


   Marker (M) bit: The marker bit indicates audioMuxElement boundaries. =
This
   bit is set to one to mark the RTP packet contains a complete
   audioMuxElement or the last fragment of an audioMuxElement.

   Timestamp: The timestamp indicates the composition time, or the
   presentation time in a no-compositor decoder. Timestamps are =
recommended
   to start at a random value for security reasons.

   Unless specified by an out-of-band means, the resolution of the =
timestamp
   is set to its default.

   Sequence Number: Increment by one for each RTP packet sent. It starts
   with a random value for security reasons.

   SSRC, CC and CSRC fields are used as described in RFC 1889 [8].

4.3 Fragmentation of MPEG-4 Audio bitstream

   It is desirable to put one audioMuxElement per RTP packet. The size =
of an
   audioMuxElement is tried to be adjusted such that the resulting RTP
   packet is not larger than the path-MTU. If this is not possible, the
   audioMuxElement MAY be fragmented across several packets based on the
   following rules.

   (1) "payloadMux" which consists of payload elements MAY be fragmented
   into several RTP packets so that one RTP packet consists of one or =
more
   payload elements. A payload element SHOULD NOT be fragmented.

   (2) If the audioMuxElement consists of the configuration information
   "setupData", the configuration information SHALL NOT be fragmented.




5. RTCP Packetization of MPEG-4 backward channel messages

   This section specifies the usage of particular RTCP packets to carry =
the
   backward channel messages generated using the MPEG-4 Audio/Visual
   backward channel messaging functionalities, e.g. NEWPRED[4]. RTCP =
packets
   specified in this section SHALL ONLY be used when it is indicated by =
the
   profile and level indication of MPEG-4 the codecs have such
   functionalities. (e.g. Advanced Real Time Simple Visual Profile[4])


   The MPEG-4 backward channel messages are transmitted on particular =
RTCP
   packets, like H.261 RTCP control packets [10].

   The MPEG-4 backward message packets differ from normal RTCP packets =
in
   that they are not transmitted to the normal RTCP destination =
transport
   address for the RTP session (which is often a multicast address).
   Instead, these backward packets are sent directly via unicast from =
the
   decoder to the coder.  The destination port for these control packets =
is
   the same port that the coder uses as a source port for transmitting =
RTP
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D


   (data) packets.  Therefore, these packets may be considered "reverse"
   control packets.

   As a consequence, these control packets may only be used when no RTP
   mixers or translators intervene in the path from the coder to the
   decoder.  If such intermediate systems do intervene, the address of
   the coder would no longer be present as the network-level source
   address in packets received by the decoder, and in fact, it might not
   be possible for the decoder to send packets directly to the coder.

   Some reliable multicast protocols use similar NACK control packets
   transmitted over the normal multicast distribution channel, however
   they typically use random delays to prevent a NACK implosion problem.
   The goal of such protocols is to provide reliable multicast packet
   delivery at the expense of delay, which is appropriate for
   applications such as a shared whiteboard.

   On the other hand, real-time Audio/Visual transmission is more =
sensitive
   to delay and does not require full reliability.  For Audio/Visual
   applications it is more effective to send the MPEG-4 backward message
   packets as soon as possible, i.e. as soon as a loss is detected, =
without
   adding any random delays.

5.1.  MPEG-4 Visual backward channel message packets definition

        0                   1                   2                   3
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |V=3D2|P|   BMT             |  PT |           length              =
|
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |                              SSRC                             |
       =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+
       |    MPEG-4 Backward Channel Message Payload (byte aligned)     |
       |                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |                               :             padding           |
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   version (V): 2 bits
        Identifies the version of RTP, which is the same in RTCP packets
        as in RTP data packets.

   padding (P): 1 bit
        If the padding bit is set, this RTCP packet contains some
        additional padding octets at the end which are not part of the
        control information. The last octet of the padding is a count of
        how many padding octets should be ignored. Padding may be needed
        by some encryption algorithms with fixed block sizes. In a
        compound RTCP packet, padding should only be required on the
        last individual packet because the compound packet is encrypted
        as a whole.

   backward channel message type (BMT): 5 bits
        Identifies the type of the MPEG-4 backward channel messages.
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D


        0:      forbidden
        1:      MPEG-4 Visual NEWPRED
        2-63:   reserved
        In this internet-draft, only NEWPRED is assigned as the =
candidate of
        the BMT for the moment.  Some other MPEG-4 Audio/Visual =
applications
        using the backward channel messages may be assigned in the =
future.

   packet type (PT): 8 bits
        The value of the packet type (PT) identifier is the constant
                for MPEG-4 backward channel messages. The assignment of =
an
   RTCP payload type for this new packet format is outside the scope of
        this document, and will not be specified here.

   SSRC: 32 bits
        SSRC is the synchronization source identifier for the sender of
        this packet.

   MPEG-4 Backward Channel Message Payload: variable
        The syntax and semantics of the MPEG-4 backward channel messages =
are
        defined in the ISO/IEC 14496-2/3[4][11]. All messages are byte
        aligned.  Normally one message is mapped onto one RTCP packet, =
and
        several messages with same BMT could be continuously mapped onto =
one
        RTCP packet.  One message SHALL NOT be fragmented into different
        RTCP packets.





6. Security Considerations

   RTP packets using the payload format defined in this specification =
are
   subject to the security considerations discussed in the RTP =
specification
   [8]. This implies that confidentiality of the media streams is =
achieved
   by encryption. Because the data compression used with this payload =
format
   is applied end-to-end, encryption may be performed on the compressed =
data
   so there is no conflict between the two operations.

   This payload type does not exhibit any significant non-uniformity in =
the
   receiver side computational complexity for packet processing  to =
cause a
   potential denial-of-service threat.


7. References


   1  Bradner, S., "The Internet Standards Process -- Revision 3", BCP =
9,
      RFC 2026, October 1996.

   2 ISO/IEC 14496-2 FDIS, _Information technology _ Generic coding of
      audio-visual objects _ Part2: Visual_, October 1998.
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D



   3 ISO/IEC 14496-3 FDIS, _Information technology _ Generic coding of
      audio-visual objects _ Part3: Audio_, October 1998.

   4 ISO/IEC 14496-2 FPDAM 1, July 1999.

   5 _Study on Internet draft for the carriage of MPEG-4 on IP_, ISO/IEC =
JTC
      1/SC 29/WG 11 N3021, October 1999.

   6 ISO/IEC 14496-1 FDIS, _Information technology _ Generic coding of
      audio-visual objects _ Part1: Systems_, October 1998.

   7  Bradner, S., "Key words for use in RFCs to Indicate Requirement
      Levels", BCP 14, RFC 2119, March 1997

   8 H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson "RTP: A =
Transport
      Protocol for Real Time Applications",  RFC 1889, Internet =
Engineering
      Task Force, January 1996.

   9 ISO/IEC JTC1/SC29/WG11/N2794, "Technical Description of the Low
      Overhead MPEG-4 Audio Transport Multiplex (LATM)", July 1999.

   10 T. Turletti, C. Hitema, "RTP Payload Format for H.261 Video =
Streams",
      RFC 2032, Octover 1996.

   11 ISO/IEC 14496-3 FPDAM 1, July 1999.




8. Author's Addresses


   Yoshihiro Kikuchi
   Toshiba corporation
   1, Komukai Toshiba-cho, Saiwai-ku, Kawasaki, 210-8582, Japan
   Email: kiku@eel.rdc.toshiba.co.jp

   Yoshinori Matsui
   Matsushita Electric Industrial Co., LTD.
   1006, Kadoma, Kadoma-shi, Osaka, Japan
   Email: matsui@drl.mei.co.jp

   Toshiyuki Nomura
   NEC Corporation
   4-1-1,Miyazaki,Miyamae-ku,Kawasaki,JAPAN
   Email: t-nomura@ccm.cl.nec.co.jp

   Shigeru Fukunaga
   Oki Electric Industry Co., Ltd.
   1-2-27 Shiromi, Chuo-ku, Osaka 540-6025 Japan.
   Email: fukunaga444@oki.co.jp
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D


   Hideaki Kimata
   Nippon Telegraph and Telephone Corporation
   1-1, Hikari-no-oka, Yokosuka-shi, Kanagawa, Japan
   Email: kimata@nttvdt.hil.ntt.co.jp

















































Kikuchi/Nomura/Fukunaga/Kimata/Matsui
=0CRTP payload format for MPEG-4 Audio/Visual streams        October =
1999=0D



Full Copyright Statement

   "Copyright (C) The Internet Society (date). All Rights Reserved.
   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implmentation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph
   are included on all such copies and derivative works. However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into





































Kikuchi/Nomura/Fukunaga/Kimata/Matsui
=0C
------=_NextPart_000_0087_01BF1EC8.9C84BAE0--




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        Omar.Elloumi@enst-bretagne.fr, opensig-announce@ctr.columbia.edu,
        osimcast@bbn.com, performance@haven.epm.ornl.gov, prs@masi.ibp.fr,
        qos-iso@mci.org.uk, qosws@dstc.edu.au, rem-conf@es.net, reres@laas.fr,
        s.low@ee.mu.oz.au, sb.all@ieee.org, sc6wg4@ntd.comsat.com,
        sig-dce@opengroup.org, sigmedia@bellcore.com, tccc@ieee.org,
        tchen@seas.smu.edu, testnet@canarie.ca, tm_member@ns.ieice.or.jp,
        xtprelay@cs.concordia.ca, lyko@kt.agh.edu.pl
Message-id: <3813DADE.5EF49F6F@ieee.org>
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***********************************
Sorry for Multiple copies of this mesage.
***********************************

Dear Colleagues:

The CFP for 3Gwireless'2000 and wmATM'2000 (the first and biggest global 3G
Wireless technical conference, with 3rd workshop on wmATM) will be due on Nov.
19, 1999. For details, pls look at the web:
http://www.itu.int/imt/5_mtgs/other_mtgs.htm, then go to "3G Wireless 2000". OR
go to: 3gwireless.com/3g/3gwireless00.

Thank you.

Sincerely,

Willie Lu
General Chair




From rem-conf-request@es.net  Mon Oct 25 06:18:02 1999
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        itc@fokus.gmd.de, itc@ieee.org, kia@usl.edu,
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        modern-heuristics@uk.ac.mailbase.ucdavis.edu,
        multicomm@cc.bellcore.com, multicomm@research.panasonic.com,
        nm-australia@amazon.postech.ac.kr, nmkorea@amazon.postech.ac.kr,
        Omar.Elloumi@enst-bretagne.fr, opensig-announce@ctr.columbia.edu,
        osimcast@bbn.com, performance@haven.epm.ornl.gov, prs@masi.ibp.fr,
        qos-iso@mci.org.uk, qosws@dstc.edu.au, rem-conf@es.net, reres@laas.fr,
        s.low@ee.mu.oz.au, sb.all@ieee.org, sc6wg4@ntd.comsat.com,
        sig-dce@opengroup.org, sigmedia@bellcore.com, tccc@ieee.org,
        tchen@seas.smu.edu, testnet@canarie.ca, tm_member@ns.ieice.or.jp,
        xtprelay@cs.concordia.ca, lyko@kt.agh.edu.pl
From: mcasoni@deis.unibo.it
Subject: Fwd: 3Gwireless'2000 and wmATM'2000
Date: Mon, 25 Oct 99 08:11:40 +0000
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Forwarded Message:

> To: cellular@comnets.rwth-aachen.de

> From: Willie LU <wwlu@ieee.org>

> Subject: 3Gwireless'2000 and wmATM'2000

> Date: Sun, 24 Oct 1999 21:21:51 -0700

> -----

<pre>
***********************************
Sorry for Multiple copies of this mesage.
***********************************

Dear Colleagues:

The CFP for 3Gwireless'2000 and wmATM'2000 (the first and biggest global 3G
Wireless technical conference, with 3rd workshop on wmATM) will be due on Nov.
19, 1999. For details, pls look at the web:
<a
href="http://www.itu.int/imt/5_mtgs/other_mtgs.htm,">http://www.itu.int/imt/5_mtgs/other_mtgs.htm,</a>then
go to "3G Wireless 2000". OR
go to: 3gwireless.com/3g/3gwireless00.

Thank you.

Sincerely,

Willie Lu
General Chair

</pre>





From rem-conf-request@es.net  Mon Oct 25 16:52:51 1999
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Sender: mathias@research.att.com
Message-ID: <3814BB90.F28FA7F5@research.att.com>
Date: Mon, 25 Oct 1999 16:20:32 -0400
From: Mathias Kretschmer <mathias@research.att.com>
Organization: AT&T Labs Research
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To: rem-conf@es.net
Subject: Updated draft: draft-ietf-avt-rtp-mpeg2aac-01.txt
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The updated draft on the MPEG-2 AAC RTP payload format is available at:

http://www.research.att.com/~mrc/draft-ietf-avt-rtp-mpeg2aac-01.txt


Comments are welcome.

-Mathias

Mathias Kretschmer, AT&T Labs Research
Room B285
180 Park Ave, Florham Park,NJ 07932
ph:  (973) 360-8899
fax  (973) 360-8455
mathias@research.att.com
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--------------ACECC18AD8824C03CDA784CF--




From rem-conf-request@es.net  Mon Oct 25 20:08:35 1999
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From: SIGNES Julien FTR&D/TD <julien.signes@rd.francetelecom.com>
To: "'Yoshihiro Kikuchi'" <kiku@eel.rdc.toshiba.co.jp>, rem-conf@es.net,
        4on2andIP-sys <4on2andIP-sys@advent.ee.columbia.edu>
Subject: RE: Submission of I/D: RTP payload format for MPEG-4 Audio/Visual
	 streams
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Hi,

I am not an MPEG audio expert, but, could anybody tell me what is the right
MPEG audio forum to inform them about our proposal to use MPEG-4 systems to
compose ITU audio streams? :)

Julien

-----Original Message-----
From: Yoshihiro Kikuchi [mailto:kiku@eel.rdc.toshiba.co.jp]
Sent: Sunday, October 24, 1999 5:09 PM
To: Yoshihiro Kikuchi; rem-conf@es.net; 4on2andIP-sys
Subject: RE: Submission of I/D: RTP payload format for MPEG-4
Audio/Visual streams


Dear all,

I'm sorry, I forgot to attach the document.

Regards,
Yoshihiro

-----Original Message-----
From: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
To: rem-conf@es.net <rem-conf@es.net>; 4on2andIP-sys
<4on2andIP-sys@advent.ee.columbia.edu>
Cc: Yoshihiro Kikuchi <kiku@eel.rdc.toshiba.co.jp>
Date: Saturday, October 23, 1999 12:34 PM
Subject: Submission of I/D: RTP payload format for MPEG-4 Audio/Visual
streams


>Dear all,
>
>I submitted an Internet draft for the RTP payload format for MPEG-4
>Audio/Visual streams:
>
>Title : RTP payload format for MPEG-4 Audio/Visual streams
>Author(s) : Yoshihiro Kikuchi, Toshiyuki Nomura, Shigeru Fukunaga,
>                 Yoshinori Matsui, Hideaki Kimata
>Filename : draft-jnb-mpeg4av-rtp-00.txt
>Pages : 14
>Date : 22-Oct-99
>
>This document describes RTP payload formats for the carriage of MPEG-4
>Audio and Visual streams, and an RTCP format for MPEG-4 backward
channel
>messages functionalities. The specification is based on the study on
>Internet draft for the carriage of MPEG-4 on IP, which is sent from
>ISO/IEC JTC1/SC29/WG11(MPEG) as a liaison statement. In this
>specification, MPEG-4 Audio/Visual bitstreams are directly mapped into
>RTP packets without using MPEG-4 Systems. The RTP header fields usage
>and the fragmentation rule for MPEG-4 Visual and Audio bitstreams are
>specified. It also specifies an RTCP packet usage to carry the MPEG-4
>backward channel messages.
>
>
>
>I attach this document. Review and comments by the experts are much
>appreciated.
>
>Best regards,
>Yoshihiro Kikuchi
>
>----
>        Yoshihiro Kikuchi
>
>E-mail: kiku@eel.rdc.toshiba.co.jp
>Corporate Research and Development Center
>TOSHIBA Corporation
>TEL: +81 +44 549 2288    FAX: +81 +44 520 1308
>
>
>

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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3DISO-2022-JP">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2448.0">
<TITLE>RE: Submission of I/D: RTP payload format for MPEG-4 =
Audio/Visual streams</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Hi,</FONT>
</P>

<P><FONT SIZE=3D2>I am not an MPEG audio expert, but, could anybody =
tell me what is the right MPEG audio forum to inform them about our =
proposal to use MPEG-4 systems to compose ITU audio streams? =
:)</FONT></P>

<P><FONT SIZE=3D2>Julien</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Yoshihiro Kikuchi [<A =
HREF=3D"mailto:kiku@eel.rdc.toshiba.co.jp">mailto:kiku@eel.rdc.toshiba.c=
o.jp</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Sunday, October 24, 1999 5:09 PM</FONT>
<BR><FONT SIZE=3D2>To: Yoshihiro Kikuchi; rem-conf@es.net; =
4on2andIP-sys</FONT>
<BR><FONT SIZE=3D2>Subject: RE: Submission of I/D: RTP payload format =
for MPEG-4</FONT>
<BR><FONT SIZE=3D2>Audio/Visual streams</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Dear all,</FONT>
</P>

<P><FONT SIZE=3D2>I'm sorry, I forgot to attach the document.</FONT>
</P>

<P><FONT SIZE=3D2>Regards,</FONT>
<BR><FONT SIZE=3D2>Yoshihiro</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Yoshihiro Kikuchi =
&lt;kiku@eel.rdc.toshiba.co.jp&gt;</FONT>
<BR><FONT SIZE=3D2>To: rem-conf@es.net &lt;rem-conf@es.net&gt;; =
4on2andIP-sys</FONT>
<BR><FONT SIZE=3D2>&lt;4on2andIP-sys@advent.ee.columbia.edu&gt;</FONT>
<BR><FONT SIZE=3D2>Cc: Yoshihiro Kikuchi =
&lt;kiku@eel.rdc.toshiba.co.jp&gt;</FONT>
<BR><FONT SIZE=3D2>Date: Saturday, October 23, 1999 12:34 PM</FONT>
<BR><FONT SIZE=3D2>Subject: Submission of I/D: RTP payload format for =
MPEG-4 Audio/Visual</FONT>
<BR><FONT SIZE=3D2>streams</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt;Dear all,</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;I submitted an Internet draft for the RTP =
payload format for MPEG-4</FONT>
<BR><FONT SIZE=3D2>&gt;Audio/Visual streams:</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;Title : RTP payload format for MPEG-4 =
Audio/Visual streams</FONT>
<BR><FONT SIZE=3D2>&gt;Author(s) : Yoshihiro Kikuchi, Toshiyuki Nomura, =
Shigeru Fukunaga,</FONT>
<BR><FONT =
SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Yoshinori Matsui, Hideaki =
Kimata</FONT>
<BR><FONT SIZE=3D2>&gt;Filename : draft-jnb-mpeg4av-rtp-00.txt</FONT>
<BR><FONT SIZE=3D2>&gt;Pages : 14</FONT>
<BR><FONT SIZE=3D2>&gt;Date : 22-Oct-99</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;This document describes RTP payload formats for =
the carriage of MPEG-4</FONT>
<BR><FONT SIZE=3D2>&gt;Audio and Visual streams, and an RTCP format for =
MPEG-4 backward</FONT>
<BR><FONT SIZE=3D2>channel</FONT>
<BR><FONT SIZE=3D2>&gt;messages functionalities. The specification is =
based on the study on</FONT>
<BR><FONT SIZE=3D2>&gt;Internet draft for the carriage of MPEG-4 on IP, =
which is sent from</FONT>
<BR><FONT SIZE=3D2>&gt;ISO/IEC JTC1/SC29/WG11(MPEG) as a liaison =
statement. In this</FONT>
<BR><FONT SIZE=3D2>&gt;specification, MPEG-4 Audio/Visual bitstreams =
are directly mapped into</FONT>
<BR><FONT SIZE=3D2>&gt;RTP packets without using MPEG-4 Systems. The =
RTP header fields usage</FONT>
<BR><FONT SIZE=3D2>&gt;and the fragmentation rule for MPEG-4 Visual and =
Audio bitstreams are</FONT>
<BR><FONT SIZE=3D2>&gt;specified. It also specifies an RTCP packet =
usage to carry the MPEG-4</FONT>
<BR><FONT SIZE=3D2>&gt;backward channel messages.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;I attach this document. Review and comments by =
the experts are much</FONT>
<BR><FONT SIZE=3D2>&gt;appreciated.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;Best regards,</FONT>
<BR><FONT SIZE=3D2>&gt;Yoshihiro Kikuchi</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;----</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Yoshihiro Kikuchi</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;E-mail: kiku@eel.rdc.toshiba.co.jp</FONT>
<BR><FONT SIZE=3D2>&gt;Corporate Research and Development Center</FONT>
<BR><FONT SIZE=3D2>&gt;TOSHIBA Corporation</FONT>
<BR><FONT SIZE=3D2>&gt;TEL: +81 +44 549 2288&nbsp;&nbsp;&nbsp; FAX: +81 =
+44 520 1308</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
</P>

</BODY>
</HTML>
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload Format for 12 bits, 20bits and 24 bits DV 
                          Audio
	Author(s)	: K. Kobayashi, A. Ogawa,  S. Casner, C. Bormann
	Filename	: draft-ietf-avt-dv-audio-00.txt
	Pages		: 10
	Date		: 25-Oct-99
	
This document specifies the packetization scheme for encapsulating
the 12 bits nonlinear, 20 bits linear and 24 bits linear audio data
streams into a payload of the Real-Time Transport Protocol (RTP).
This Internet draft is a revised draft we submitted name as 'draft-
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Thank you for asking.
The latest version does seem to be an improvement.  I especially like the
end or 'E' bit.

I still have some concerns and questions though, to wit:

Paragraph 2 of section 3.1 (Intro to RTP Payload Format for Named Telephone
Events):
states that "DTMF digits ... SHOULD use the same sequence number and
time-stamp base as the regular audio channel".
It strikes me that a receiver might use VERY different strategies for
processing these events depending on whether the sequence number is
the same as the regular audio or not.  This basic fact would need to
somehow be negotiated by out of band means.  I think this is too much
wiggle
room and would argue for MUST instead of SHOULD.   Alternatively, some
discussion of how SDP or other techniques could be used to negotiate
or declare this should be inserted.
If, for example, one is not taking advantage of redundancy then detecting a
missing DTMF packet which is isolated or in the beginning of a sequence
is impossible if the same sequence number as the regular audio channel is
not used because you don't know when to expect it (of course you know when
the next one arrives with the sequence number bumped up by two but that is
too late).  If this concern is correct, then I think it merits at the very
least a cautionary statement
in the document.
There are also some issues related to compound packets and jitter buffers
raised by having a different sequence of sequence numbers/timestamps.
One might need to maintain two parallel buffers.

In section 3.5 on the 'E' bit is the sentence:
"A sender MAY set the end bit only when retransmitting the last packet for
a tone. This avoids having to wait to detect whether the tone has indeed
ended."
The second sentence seems to suggest that the authors meant to say MUST
instead of MAY, and that the emphasis is on last and not retransmit.  The
sentence
is sort of ambiguous.
In any case, it seems to me most useful to set the E bit for each of the
transmissions and retransmissions of the last packet for a tone.  This
affords maximum protection for detecting the end in the event of packet
loss. Avoiding the wait would best be acheived by setting the end bit on
the first transmission of the last packet, not the retransmission (If I'm
following this correctly.)
(There is also some kind of typo in the first sentence of the 2nd paragraph
for the 'E' bit where it says: "Receiver implementations MAY use at
different ....")

In the example in section 3.8, I believe the E bit has been left out and
two R bits are still shown.
Translating between milliseconds and timestamp units is helpful, it would
be more helpful still to point out things like 4800 = 11200 - 6400  right
in the
timestamp offset field for the second digit.





From rem-conf-request@es.net  Tue Oct 26 13:19:57 1999
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From: Florissa Colina <florissa@bmrc.berkeley.edu>
Subject: 10/27 Supporting Multicast Services in the Emerging Internet2
  Network Infrastructure The View from Minus Six Feet  -- Ken Lindahl,
  Communication and Network Services U.C. Berkeley 
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Berkeley Multimedia, Graphics and Interfaces Seminar

Supporting Multicast Services in the Emerging Internet2 Network
Infrastructure The View from Minus Six Feet

Wednesday October 27, 1999, 
1:00-2:30 p.m. PDT
Fujitsu Seminar Room (405 Soda Hall) 


Ken Lindahl 
Communication and Network Services U.C. Berkeley  

IP multicast is recognized by the Internet2 Project to be an important
technology for the delivery of education and other services in future
advanced communication infrastructure. The I2 Multicast Working Group was
created to encourage the deployment of "a consistent and ubiquitous
multicast service within the Internet2 community." 

In this talk, I will describe the current state of multicast routing in the
high-performance "next generation" backbones, vBNS and Abilene; in
California's advanced educational network, CalREN-2, and in the UC Berkeley
campus network. I'll mention some successes and difficulties encountered in
trying to deploy and support multicast service, as well as our attempts to
implement the recommendations and results of activies in the I2 Multicast
Working Group and the IETF MBONE Deployment (mboned) Working Group.
---------------

The seminar is broadcast live on the Internet Mbone and as a Real
Networks G2 broadcast.

You can connect to the live RealNetworks broadcast at:

http://media2.bmrc.berkeley.edu/bibs/schedule.cfm 
You'll see a link to cs298-5/real networks/live programs.  

You can also directly put in this url into the Real Player:

rtsp://media2.bmrc.berkeley.edu/encoder/cs298-5.rm

To do so you will need to have the "Real Player G2" installed, which is
available from:
   http://www.real.com/products/player/downloadrealplayer.html

To tune into the Internet MBone broadcast, look for the announcement in
your MBone session directory program ('sdr').  If you are not receiving
the announcement you may be able to receive the session by manually
configuring the client programs ('vic', and 'vat') with the session
addresses:

low bit rate
	video: 233.0.25.1/22334
	audio: 233.0.25.2/22446

medium bit rate
	video: 233.0.25.129/22334
	audio: 233.0.25.2/22446

You can get further information about this seminar, and access to
replays of previous seminars at the MIG Seminar web page:

http://bmrc.berkeley.edu/298/index.html

Sponsored by the Berkeley Multimedia Research Center 




From rem-conf-request@es.net  Tue Oct 26 15:20:04 1999
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Date: Tue, 26 Oct 1999 15:04:07 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Bruce_Levens@nmss.com wrote:
> 

> Paragraph 2 of section 3.1 (Intro to RTP Payload Format for Named Telephone
> Events):
> states that "DTMF digits ... SHOULD use the same sequence number and
> time-stamp base as the regular audio channel".
> It strikes me that a receiver might use VERY different strategies for
> processing these events depending on whether the sequence number is
> the same as the regular audio or not.  This basic fact would need to
> somehow be negotiated by out of band means.  I think this is too much
> wiggle
> room and would argue for MUST instead of SHOULD.   

Given that nobody has indicated a valid reason why one should use a
different timestamp base (and that switching encoders, say, between
G.711 and G.723.1 wouldn't change the sequence number either), I'd agree
that requiring this simplifies life. Changed, unless I hear a
counter-argument.


> 
> In section 3.5 on the 'E' bit is the sentence:
> "A sender MAY set the end bit only when retransmitting the last packet for
> a tone. This avoids having to wait to detect whether the tone has indeed
> ended."

I've reworded the sentence to

"A sender {\MAY} delay setting the end bit until retransmitting the last
packet for a tone, rather than on its first transmission.  This avoids
having to wait to detect whether the tone has indeed ended."


> In any case, it seems to me most useful to set the E bit for each of the
> transmissions and retransmissions of the last packet for a tone.  This
> affords maximum protection for detecting the end in the event of packet
> loss. Avoiding the wait would best be acheived by setting the end bit on
> the first transmission of the last packet, not the retransmission (If I'm
> following this correctly.)

Yes, from a receiver perspective. But a sender wouldn't necessarily know
whether the tone has ended when it sends the packet, as it would have to
look into the future for that. It is rather unlikely that the tone will
end just after the packet interval, but it could conceivably happen. The
problem would not exist if tones were truly continuous, but since the
detector has to "bridge" short gaps, it can't decide whether the short
gap at the end of the interval is indeed just a spike and should be
ignored or the end of the tone was reached. In ASCII art:

       | transmission point
xxxxx  |   xxxx              - tone with small gap
xxxxx  |                       end of tone     


> In the example in section 3.8, I believe the E bit has been left out and
> two R bits are still shown.

Forgot to correct the example; fixed.

> Translating between milliseconds and timestamp units is helpful, it would
> be more helpful still to point out things like 4800 = 11200 - 6400  right
> in the
> timestamp offset field for the second digit.

I added the calculation in the figure.

Thanks for your comments.

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload Format for 12 bits, 20bits and 24 bits DV 
                          Audio
	Author(s)	: K. Kobayashi, A. Ogawa,  S. Casner, C. Bormann
	Filename	: draft-ietf-avt-dv-audio-00.txt
	Pages		: 10
	Date		: 25-Oct-99
	
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the USA and overseas, including the publishers themselves!   It is also
*highly unlikely* you will find any of their USA competitors calling you
overseas, in order to offer that personal touch, just to sell you a couple
of magazines!  But that is what this company specializes in and loves
doing!     Around one-half their business comes from overseas, so they are
very patient with new customers who only speak limited English as a 2nd
language.    Subscription prices quoted for overseas consist of the
subscription price, plus the FPH.   You add the two together and that is
your total cost.   The exception is the 1 yr. freebie you get with your
first order.   On that title, you pay *only* the FPH for the 1 yr. term.

Their prices are so cheap because when you deal with them, you cut-out all
the middlemen.


HERE IS HOW YOU CAN GET MORE INFO AND GET STARTED WITH THEM:

Simply fax or smail back to us the reply form listed at the top of this
message.   We will then forward your form on to the subscription agency. 
They will then email their "big and juicy" catalogue to you, in whichever
of the two formats you chose.   The catalogue is FREE and makes for hours
of fascinating reading, on its own. It includes the complete list of
freebies, a complete list of all the titles they sell, as well as detailed
descriptions on most of the titles, along with lists of titles by category
of interest and their terms of sale.    

They will then give you a friendly, no-pressure, no obligation, 5-minute
call to go over how they work and to answer any questions that you might
have, as well as give you up-to-the minute price quotes on any titles you
might be considering.     They will call you in whatever country you live
in, taking the time difference into account.        As they like to
emphasize the personal touch they give to each new customer, all first-time
orders can only be done via phone, so they can answer all your questions
completely and personally.   Once you have placed your first order via
phone, you will be able to place future orders and make inquiries on your
account, get price quotes, etc., all via email, if that is most convenient
for you.

Within the USA, they accept payment via check over the phone, Mastercard,
Visa, American Express, Diner's Club and Carte Blanche.    Overseas, they
accept Mastercard, Visa, American Express, Diner's Club and Carte Blanche,
even if your credit card is a local one in local currency (that most
merchants in the USA would not normally be willing to accept).

That's our introduction of our client that we represent.   We hope that we
have piqued your interest and that you will take the next step to get their
free catalogue!   Thank you for your time and interest.

--
Tempting Tear-Outs.
For more info on marketing & consulting rates, please write us on your
company letterhead, w/business card, via smail to:   Tempting Tear-Outs,
PMB 200, 3835 Richmond Ave., Staten Island NY  10312-3828, USA.     

This email message has been sent to you by:  Tempting Tear-Outs, PMB 200,
3835 Richmond Ave., Staten Island NY  10312-3828, USA..     


TO BE REMOVED FROM OUR MAILING LIST:

There are 4 easy methods to let us know that you would like to be removed
from our mailing list (please use whichever way is most convenient for you;
 if you are unable to reach us via your chosen method, please try another
of the methods below, in order that we can remove you from our mailing
list):

1. 	email us at our "from" email address at the top of this message, with a
subject of 	"remove" and a blank body OR

2. 	fax us a 1-line message at:    1-435-302-5907* ;   the 1-line message
should say:  	"Remove  	________@_________   (your email address)" OR

3.	leave us a 1-sentence voicemail message at:    1-435-302-5907* ;   the
1-sentence 	message should say:  	"Remove  	________@_________   (your
email 	address)";  please speak clearly and spell out your email address
phonetically OR

4. 	mail us a 1-line message at:   Tempting Tear-Outs, PMB 200, 3835
Richmond Ave., 
      	Staten Island NY  10312-3828;  the 1-line message should say:
	"Remove  ________@_________   (your email address)."

*Please note:  1-435-302-5907 is ONLY for remove requests.   Requests for
more information forms sent to 1-435-302-5907 CANNOT be acknowledged.   For
more information, please use the fax numbers listed earlier above the reply
form, following the directions very carefully.      




From rem-conf-request@es.net  Wed Oct 27 03:55:31 1999
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Date: Wed, 27 Oct 1999 16:46:08 +0900
From: Murthy Nanduri <nvnmurthy@seeyes.co.in>
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Subject: Padding in RTP
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Hi,

I am working on the implementation of RFC2508.i.e., the
RTP/UDP/IP Header compression and decompression.

I have tested my implementation, with an equipment which
sends the RTP packets with the Marker bit and Padding bits
NOT set. The implementation is working fine and I could able
to hear the voice properly between the two machines.

When I started testing the CRTP implementation with another
equipment which sends the RTP packets with the Marker bit and
Padding bits SET(for some packets of stream), I am unable to hear the
voice between the two machines.

As per the RFC2508, I am setting the Bitmask for the Marker bit while
sending the compressed RTP packet.

When the padding bit is set, I am sending the compressed UDP packet..

Can you please let me know what has to be done by CRTP
with the padding bytes?

I want to know whether the padding bytes have to be removed
by CRTP compressor/decompressor OR it is the responsibility
of the RTP applications to ignore the padding bytes..

I think it is the responsibility of the Voice applications to take care
of the
padding bytes.. Am I right?

can U please let me know what for the Marker bit and Padding bits are
used?

Murthy NVN







From rem-conf-request@es.net  Wed Oct 27 07:07:23 1999
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To: Stephen Casner <casner@cisco.com>
cc: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>, rem-conf@es.net
Subject: Re: RTP spec questions
In-Reply-To: Message from Stephen Casner <casner@cisco.com> of "Thu, 21 Oct 1999 21:52:23 PDT." <Pine.WNT.4.10.9910212141310.517-100000@casner-pc.cisco.com>
Date: Wed, 27 Oct 1999 11:51:38 +0100
From: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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--> Stephen Casner writes:
>On Fri, 22 Oct 1999, Jonathan Rosenberg wrote:
>> > A related question has occurred to me just recently:  If an RTCP
>> > compound packet includes information from more than one host, as in
>> > the case where an RTP mixer aggregates SDES information from multiple
>> > sources, should the size of that packet be divided by the number of
>> > sources before being figured into the average?
>> 
>> Good catch; I believe it does need to be divided in this way. Otherwise,
>> if the packet size is more than 5 times the average size, there will be
>> constant timeouts.
>
>I'm not sure that's true.  Everyone should be averaging over the same
>set of packets, so the intervals should be roughly consistent.  Well,
>execept that the estimator gain is 1/16 so it could bounce around if
>the big packet was only one out of many.  My concern was more that the
>average would be wrong -- too large, which means intervals longer than
>they should be, which is the safer direction.

I'd be more tempted to follow Jonathan's other suggestion, and allow a
mixer which is mixing N sources to use N times the single source RTCP
bandwidth. This would also let you get away with the naive implementation
of sending an RTCP packet on the mixed side each time a packet is received
on the non-mixed side (ie: rewrite the SR/RR but forward SDES packets as
soon as they are received).

Colin



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Interoperability Statement
	Author(s)	: C. Perkins
	Filename	: draft-ietf-avt-rtp-interop-02.txt
	Pages		: 6
	Date		: 26-Oct-99
	
It is required to demonstrate interoperability of RTP implementations
in order to move the RTP specification to draft standard.  This memo
outlines those features to be tested, as the first stage of an
interoperability statement.

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Testing Strategies
	Author(s)	: C. Perkins, J. Rosenberg, H. Schulzrinne
 	Filename	: draft-ietf-avt-rtptest-02.txt
	Pages		: 19
	Date		: 26-Oct-99
	
This memo describes a possible testing strategy for RTP implementations.
The tests are intended to help demonstrate interoperability of multiple
implementations, and to illustrate common implementation errors.  They
are not intended to be an exhaustive set of tests and passing these
tests does not necessarily imply conformance to the complete RTP
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From: icme2000@penguin.poly.edu
Subject: 	IEEE INTERNATIONAL CONFERENCE ON MULTIMEDIA AND EXPO
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	IEEE INTERNATIONAL CONFERENCE ON MULTIMEDIA AND EXPO

	       JULY 30 -- AUGUST 2, 2000, NEW YORK CITY

	        http://www.ieee-multimedia.poly.edu


-----------------CALL FOR PAPERS--------------------

International Conference on Multimedia & Expo 2000 is the first IEEE
multimedia conference that will take place in joint collaboration with
four IEEE societies. The same four IEEE societies that have recently
launched the IEEE Transactions on Multimedia, namely the Circuits and
Systems Society, the Computer Society, the Signal Processing Society
and the Communications Society have again come together in making this
one-of-a-kind conference a reality.

The goal is to unify all IEEE as well as non-IEEE multimedia related
activities under the common umbrella of ICME 2000 -- the millennium
multimedia meeting. It is expected that this will become a yearly
event as the flagship Multimedia conference of the IEEE.  New York
City (the Big Apple) is uniquely suited for this first-time event in
view of its large concentration of high tech Industries, first-rate
academic institutions and numerous small multimedia companies. The
intent is to blend the traditional IEEE conference framework into an
Expo by providing the industry an opportunity to showcase products.

Authors should submit a four-page manuscript in double-column format
including authors' names, affiliations and a short abstract. Only
electronic submission (in postscript format) will be accepted.
Electronic submission procedure will be made available at the
conference web site:

	http://www.ieee-multimedia.poly.edu

Topics covered include but are not limited to the following:
	* Signal processing for media integration
	* Components and technologies for multimedia systems
	* Human-machine interface and interaction
	* Multimedia databases and file systems
	* Multimedia communication and networking
	* System integration
	* Applications
	* STDS standards and related issues

Once accepted, the authors will be asked to prepare a four-page camera ready
paper to appear in the conference proceedings and CD-ROM proceedings.

Proposals for special sessions and half-day or full-day tutorial courses
may be submitted to the respective chairs before the respective
deadlines. Visit the conference web site for further information.


			DEADLINES

	December 1, 1999	Submission of regular papers
	February 1, 2000	Proposals for special sessions, panels, tutorials, demos
	March 1, 2000	Notification of acceptance for papers
	March 15, 2000	Notification of acceptance for proposals
	April 1, 2000	Paper camera-ready copy due




		CONFERENCE COMMITTEE

	General Chair:
		Sankar Basu, IBM
		sankar@watson.ibm.com

	Finance Chair:
		Peter Westerink, IBM

	Technical Program Co-Chairs:
		Tsuhan Chen, Carnegie Mellon
		Jeff H. Derby, IBM
		Peiya Liu, Siemens
		Chung-Sheng Li, IBM

	Special Sessions:
		Raj Acharya, SUNY, Buffalo
		Chalapathy Neti, IBM

	Tutorials:
		Rama Chellappa, U. Maryland
		Alexandros Eleftheriadis, Columbia

	Plenaries:
		S. Y. Kung, Princeton U.
		Shih-Fu Chang, Columbia U.

	Demos/Exhibits:
		Yao Wang, NY Polytechnic U.

	Publicity:
		Jelena Kovacevic, Bell Labs
		Ed Wong, NY Polytechnic

	Publications:
		Y. Q. Shi, NJIT
		Bhuvana Ramabhadran, IBM

	Registration:
		Junehwa Song, IBM

	Local Arrangements:
		Goran Djuknic, Lucent Tech.
		Ravi Rao, IBM
		Atul Puri, AT&T


      Electronic Media:
           Raj Kumar Rajendran, Columbia University
           Nasir Memon, Polytechnic University

	European Liaison:
		Alberto Del Bimbo, U. Florence
		Martin Vetterli, EPFL

	Far East Liaison:
		Masahito Hirakawa, Hiroshima U.
		Sadaoki Furui, Tokyo Inst. Tech.

	Steering Committee:
		Tadao Ichikawa, Hiroshima U.
		Rama Chellappa, U. Maryland
		Gary W. Cobb, Dell
		Alberto Del Bimbo, U. Florence
		Jeff H. Derby, IBM
		Alexander D. Gelman, Panasonic
		Yih-Fang Huang, U. Notre Dame
		Chung-Sheng Li, IBM
		Peiya Liu, Siemens
		Antonio Ortega, USC
		Dipankar Raychaudhuri, NEC
		Bing Sheu, Avanti Corporation
		John A. Sorensen, TU Denmark

	Advisory Committee:
		Richard Cox, AT&T
		Ed Deprettere, Delft U. Tech.
		Cesar Gonzales, IBM
		Scott Kirkpatrick, IBM
		Michael Picheny, IBM








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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload Format for MPEG-4 Streams
	Author(s)	: R. Civanlar,  A. Basso, S. Casner, C. Herpel,
                          C. Perkins
	Filename	: draft-ietf-avt-rtp-mpeg4-02.txt,.ps
	Pages		: 10
	Date		: 26-Oct-99
	
This document describes a payload format for transporting MPEG-4
encoded data using RTP. MPEG-4 is a recent standard from ISO/IEC for
the coding of natural and synthetic audio-visual data. Several
services provided by RTP are beneficial for MPEG-4 encoded data
transport over the Internet. Additionally, the use of RTP makes it
possible to synchronize MPEG-4 data with other real-time data types.
This specification is a product of the Audio/Video Transport working
group within the Internet Engineering Task Force and ISO/IEC MPEG-4 ad
hoc group on MPEG-4 over Internet. Comments are solicited and should
be addressed to the working group's mailing list at rem-conf@es.net
and/or the authors.

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From rem-conf-request@es.net  Wed Oct 27 10:38:53 1999
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From: Parekh Rishabh-crp049 <R.Parekh@motorola.com>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Cc: "Mike Korus (E-mail)" <cmk013@email.mot.com>
Subject: RTP port range question
Date: Wed, 27 Oct 1999 09:16:20 -0500
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I have a question regarding the UDP ports used for RTP. Here is an excerpt
from RFC 1889.

"RTP relies on the underlying protocol(s) to provide demultiplexing of RTP
data and RTCP control streams. For UDP and similar protocols, RTP uses an
even port number and the corresponding RTCP stream uses the next higher
(odd) port number. If an application is supplied with an
odd number for use as the RTP port, it should replace this number with the
next lower (even) number."

Does RTP use an even UDP source port or an even UDP destination port? 

Thanks to all who reply,
Rishabh Parekh
Motorola



From rem-conf-request@es.net  Wed Oct 27 11:33:34 1999
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Date: Wed, 27 Oct 1999 11:21:40 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: Dynamicsoft
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To: Parekh Rishabh-crp049 <R.Parekh@motorola.com>
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Parekh Rishabh-crp049 wrote:
> 
> I have a question regarding the UDP ports used for RTP. Here is an excerpt
> from RFC 1889.
> 
> "RTP relies on the underlying protocol(s) to provide demultiplexing of RTP
> data and RTCP control streams. For UDP and similar protocols, RTP uses an
> even port number and the corresponding RTCP stream uses the next higher
> (odd) port number. If an application is supplied with an
> odd number for use as the RTP port, it should replace this number with the
> next lower (even) number."
> 
> Does RTP use an even UDP source port or an even UDP destination port?

The source port is irrelevant. Its the destination port.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 210 
Dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From rem-conf-request@es.net  Thu Oct 28 03:21:18 1999
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Date: Wed, 27 Oct 1999 23:55:16 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: rem-conf@es.net
cc: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
Subject: Working group last call on RTP to Draft Standard
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To the AVT Working Group:

This message announces the working group last call for comments on the
set of drafts needed to advance RTP spec to Draft Standard status.
This working group last call is to end with the AVT session at the
IETF meeting in two weeks.  At the meeting, we will propose submission
for IESG Last Call.  Assuming there are no objections, the goal will
be to make edits on site if possible to address any comments, and then
submit to the Area Directors.  We'd like to conclude this process!

In addition, this is a request to RTP implementors to fill in the
interoperability checklist that we will need to accompany the drafts
for advancement to Draft Standard.  We've made the same request
informally before, but this time we're asking that you do this in the
next two weeks so that we can have at least a first-pass tally at the
meeting.  If it looks like there are features for which we don't have
two interoperable implementations, we may have to remove those
features from the draft, or do some additional implementation and
interop testing!

Here are the details on the drafts starting WG last call:

The RTP spec and the A/V Profile have been updated to reflect the
changes agreed upon at the last meeting.  These drafts have not made
it through the Internet-Drafts queue yet, so you can get them from

    ftp://ftpeng.cisco.com/casner/outgoing/draft-ietf-avt-rtp-new-05.ps
    ftp://ftpeng.cisco.com/casner/outgoing/draft-ietf-avt-rtp-new-05.txt
    ftp://ftpeng.cisco.com/casner/outgoing/draft-ietf-avt-profile-new-07.ps
    ftp://ftpeng.cisco.com/casner/outgoing/draft-ietf-avt-profile-new-07.txt

The .ps versions have changebars relative to RFC 1889 and 1890.

The spec and profile drafts need to be accompanied by the drafts on
RTP MIME type registration and SDP bandwidth specifiers for RTCP which
are referenced (non-normatively) by the spec and profile drafts.
These drafts will be submitted for publication as Proposed Standards.

    ftp://ftpeng.cisco.com/casner/outgoing/draft-ietf-avt-rtp-mime-01.txt
    draft-ietf-avt-rtcp-bw-00.txt  (no recent edits)

RTP testing and interoperability are addressed in Informational drafts
that have also been updated and should appear shortly:

    draft-ietf-avt-rtptest-02.txt
    draft-ietf-avt-rtp-interop-02.txt

The second of these drafts is the checklist for interoperability
statements on the RTP spec.  We do need one more draft to cover the
interoperability criteria for the A/V profile, which is to be posted
shortly after the IETF meeting.
					-- Steve and Colin




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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: Multiplexing Scheme for RTP Flows between Access 
                          Routers
	Author(s)	: K. El-khatib, G. Luo,  G. Bochmann, P. Feng
	Filename	: draft-ietf-avt-multiplexing-rtp-01.txt
	Pages		: 13
	Date		: 27-Oct-99
	
This draft proposes a light-weight data driven approach for 
multiplexing low bit rate RTP streams at the edge router of the 
Internet in order to reduce the RTP/UDP/IP header overhead associated 
with each RTP stream. Audio packets from different sources in a local 
access network destined to different users in the same remote access 
network are multiplexed into one packet, with the original RTP/UDP/IP 
header of each packet replaced with a mini-header (2 bytes), resulting 
in a reduction of the overhead. Access routers can use the IP telephony 
Border Gateway Protocol (TBGP) to exchange the reachability of IP 
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Profile for Audio and Video Conferences with 
                          Minimal Control
	Author(s)	: H. Schulzrinne, S. Casner
	Filename	: draft-ietf-avt-profile-new-07.txt,.ps
	Pages		: 37
	Date		: 27-Oct-99
	
This memorandum is a revision of RFC 1890 in preparation for
advancement from Proposed Standard to Draft Standard status. Readers
are encouraged to use the PostScript form of this draft to see where
changes from RFC 1890 are marked by change bars.
This document describes a profile called 'RTP/AVP' for the use of the
real-time transport protocol (RTP), version 2, and the associated
control protocol, RTCP, within audio and video multiparticipant
conferences with minimal control. It provides interpretations of
generic fields within the RTP specification suitable for audio and
video conferences. In particular, this document defines a set of
default mappings from payload type numbers to encodings.
This document also describes how audio and video data may be carried
within RTP. It defines a set of standard encodings and their names
when used within RTP. The descriptions provide pointers to reference
implementations and the detailed standards. This document is meant as
an aid for implementors of audio, video and other real-time
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: Real-Time Transport Protocol Management Information 
                          Base
	Author(s)	: M. Baugher, I. Suconick,B. Strahm
	Filename	: draft-ietf-avt-rtp-mib-07.txt
	Pages		: 37
	Date		: 27-Oct-99
	
This memo defines a portion of the Management Information Base
(MIB) for use with network management protocols in the Internet
community.  In particular, it defines objects for managing Real-Time 
Transport Protocol(RTP) systems [RFC1889].

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: MIME Type Registration of RTP Payload Formats
	Author(s)	: S. Casner, P. Hoschka
	Filename	: draft-ietf-avt-rtp-mime-01.txt
	Pages		: 18
	Date		: 27-Oct-99
	
This document defines the procedure to register RTP Payload Formats
as audio or video MIME subtype names.  This is useful in a text-
based format or control protocol to identify the type of an RTP
transmission.  This document also registers all the RTP payload
formats defined in the RTP Profile for Audio and Video Conferences
as MIME subtypes.  Some of these may also be used for transfer
modes other than RTP.

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP: A Transport Protocol for Real-Time Applications
	Author(s)	: H. Schulzrinne,  S. Casner,  R. Frederick, 
                          V. Jacobson,
	Filename	: draft-ietf-avt-rtp-new-05.txt,.ps
	Pages		: 100
	Date		: 27-Oct-99
	
This memorandum is a revision of RFC 1889 in preparation for
advancement from Proposed Standard to Draft Standard status. Readers
are encouraged to use the PostScript form of this draft to see where
changes from RFC 1889 are marked by change bars.
This memorandum describes RTP, the real-time transport protocol. RTP
provides end-to-end network transport functions suitable for
applications transmitting real-time data, such as audio, video or
simulation data, over multicast or unicast network services. RTP does
not address resource reservation and does not guarantee quality-of-
service for real-time services. The data transport is augmented by a
control protocol (RTCP) to allow monitoring of the data delivery in a
manner scalable to large multicast networks, and to provide minimal
control and identification functionality. RTP and RTCP are designed
to be independent of the underlying transport and network layers. The
protocol supports the use of RTP-level translators and mixers.
This specification is a product of the Audio/Video Transport working
group within the Internet Engineering Task Force. Comments are
solicited and should be addressed to the working group's mailing list
at rem-conf@es.net and/or the authors.

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From rem-conf-request@es.net  Thu Oct 28 07:38:28 1999
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A New Internet-Draft is available from the on-line Internet-Drafts
directories.  This draft is a work item of the Audio/Video Transport
Working Group of the IETF.

	Title           : RTP Payload for DTMF Digits, Telephony Tones
			  and Telephony Signals
	Author(s)	: H. Schulzrinne, S. Petrack
	Filename	: draft-ietf-avt-tones-02.txt,.ps
	Pages		: 28
	Date		: 27-Oct-99
	
This memo describes how to carry dual-tone multifrequency (DTMF)
signaling, other tone signals and telephony events in RTP packets.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-tones-02.txt

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	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
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	"multipart" MIME messages (i.e. documents which have been split
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From rem-conf-request@es.net  Thu Oct 28 14:11:42 1999
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          id <g.13674-0@bells.cs.ucl.ac.uk>; Thu, 28 Oct 1999 17:37:52 +0100
To: rem-conf@es.net
Subject: Outline AVT agenda
cc: casner@cisco.com
Date: Thu, 28 Oct 1999 17:37:52 +0100
Message-ID: <3702.941128672@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Folks,

Enclosed is the list of requests for agenda time I have received for the
AVT meeting at the Washington DC IETF. If there are any additional items
or corrections, please contact me as soon as possible. I will be putting
the final agenda together early next week.

Thanks,
Colin

===============================================================================
- Introduction and status update
	draft-ietf-avt-fec-08.txt
	draft-ietf-avt-rtp-format-guidelines-04.txt 
	draft-ietf-avt-rtpsample-05.txt
	draft-ietf-avt-rtp-mib-07.txt
	draft-ietf-avt-tones-01.txt

- RTP and the audio/video profile
	draft-ietf-avt-rtp-new-05.txt
	draft-ietf-avt-profile-new-06.txt
	draft-ietf-avt-rtp-mime-00.txt
	draft-ietf-avt-rtcp-bw-00.txt
	draft-ietf-avt-rtp-interop-02.txt
	draft-ietf-avt-rtptest-02.txt

- RTP Payload for text conversation		
	draft-ietf-avt-rtp-text-02			(Hellstrom)

- RTP Payload Format for DV Format Video
	draft-ietf-avt-dv-video-01.txt			(Kobayashi)

- RTP Payload Format for 12 bits, 20bits and 24 bits DV Audio
	draft-ietf-avt-dv-audio-00.txt			(Kobayashi)

- RTP Payload Format for MPEG-2 AAC Streams
	draft-ietf-avt-rtp-mpeg2aac-01.txt		(Basso)

- A More Loss-Tolerant RTP Payload Format for MP3 Audio
	draft-ietf-avt-rtp-mp3-01.txt			(Finlayson)

- Transport of MPEG-4
	draft-ietf-avt-rtp-mpeg4-02.txt			(Basso)
	draft-guillemot-genrtp-01.txt			(Christ)
	draft-jnb-mpeg4av-rtp-00.txt			(Harasaki)
	draft-singer-rtp-qtfile-01.txt			(Singer)
	Liason statement from MPEG (29n3290.zip)

- An RTCP-based Retransmission Protocol for Unicast RTP Streaming Multimedia
	draft-podolsky-avt-rtprx-00.txt 		(Yano)

- Dynamic Nx64 trunking
	draft-tulai-avt-dynamic-nx64-00.txt		(Tulai)
===============================================================================



From rem-conf-request@es.net  Thu Oct 28 14:15:08 1999
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Message-ID: <381888D7.A10E0EC4@cs.columbia.edu>
Date: Thu, 28 Oct 1999 13:33:11 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
X-Mailer: Mozilla 4.61 [en] (X11; U; SunOS 5.7 sun4u)
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To: rem-conf@es.net
Subject: Voice activity detection software
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I'm looking for voice-activity detection software modules. I'm aware of
the modules in vat, nevot and rat. Anything else?
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf-request@es.net  Thu Oct 28 22:58:09 1999
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From: Humphrey Liu <hcliu@cisco.com>
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Dear all,

I am submitting a draft of delivering MPEG2 multiple program transport stream (MPTS) for consideration. Essentially, this draft is the extension of RFC2250. The reason why this extension is needed is that when the encapsulated MPEG2 MPTS has high bandwidth such as 30-60Mbps, the original base clock, 90KHz, defined in RFC2250, is not good enough. Thus, I propose in this draft to extend the clock frequency to 27MHz, the same frequency as MPEG2 Program Clock Reference.

I understand that I missed the submission deadline for the upcoming meeting.  However, I wonder if there is any chance that the draft can be still included for discussion. I certainly would like to present this draft to the working group to moving forward.

Any help will be highly appreciately!!

Humphrey Liu
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Internet Engineering Task Force	             Audio/Video Transport Working Group
INTERNET-DRAFT                                                            H. Liu
draft-ietf-avt-rtp-mp2t-00.txt	                                   Cisco Systems

                                                                October 22, 1999
                                                             Expires: 04/22/2000


    Extension of RTP payload Type for Multiple Program MPEG Transport Stream



Abstract

This document is to define a dynamic payload type that extends the payload type 
of 33, defined in RFC1890 [4] for MPEG2 transport streams. The usage of payload
type of 33 is defined in RFC2250 [1]. This purpose of this extension is to 
enhance the RTP protocol in such way that it can be used to deliver multiple 
program MPEG transport stream as well.


1. Introduction

   Recent effort in the DVB and ATSC standard body has endorsed MPEG2 transport 
stream format as the broadcast transport standard.  With appropriate internet 
standard such as RFC 2236 [2] for multicast and RFC 2205 [3] and other RFC's 
for QoS, real-time streaming becomes quite feasible. Also with the technology 
advances on the "last-mile" network such as cable modem and DSL, it is quite 
possible to deliver high bandwidth, broadcast quality, and individualized MPEG 
video directly to home.

Section 2 in RFC2250 defines how to encapsulate MPEG2 transport stream into RTP 
packet format. RFC2250 defines the base frequency for MPEG2 transport stream
to be 90KHz. While this base clock is proper for an MPEG single program 
transport stream (SPTS), it losses the precision for MPEG multiple program 
transport stream (MPTS). Thus, a new payload type is needed when RTP is used to
transport MPEG MPTS.


2. Extension to Payload Type for MP2T

ISO/IEC 13818-1 [5] defines MPEG transport packet format to be able to carry 
not only video, audio, but also any program related data. Further more, with 
ISO/IEC 13818-6, DSM-CC [6], and the Radio Frequency Interface Specification of 
Data-Over-Cable Services Interface Specification (DOCSIS), defined by 
CableLab [7], the same packet format can be used to deliver IP data as well. 

According to ISO/IEC 13818-1, at MPEG transport packet layer, every transport 
packet is a sample unit in time, instead of the video or audio access units. 
Thus, the sample clock frequency should be suitable for the packet samples. In 
the satellite and cable industry, each video frequency carrier typically 
contains from 27Mbps to 50Mbps.  Each frequency carrier can contain up to 
40,000 MPEG transport packets. If the base clock frequency 90,000 defined by 
the payload type of 33 is to be used, each packet only has 2.5 clock tick time. 
Any round-off error can count for 30% error rate. Thus, the existing payload 
type 33 is not suitable to deliver MPEG MPTS. 

Furthermore, if payload type of 33 is to be used for SPTS, one can argue that, 
in this case, PCR and continuity counter embedded in the MPEG transport packet 
header can be used for the purpose of clock synchronization and packet drop 
detection, repectively. In fact, several vendors in the MPEG industry have 
selected UDP/IP for transporting MPEG2 SPTS in the IP network, instead of RTP. 
Thus, the most compelling application for the RTP payload type of 33 is for 
transporting MPEG MPTS. Because of that, the base clock with higher frequency 
should be defined to allow a reasonable RTP deliver scheme for MPEG MPTS.

This document defines a new dynamic payload type of 96 that extends the base 
clock frequency payload type of 33 from 90KHz to 27MHz.  27MHz is the same as 
the Programe Clock Reference (PCR) defined in ISO/IEC 13818-1.  Now that MPEG 
MPTS is to feed any NTSC/PAL receiver, the clock recovery has to be very 
conservative. The same requirement on the clock frequency and clock recovery 
rate from ISO/IEC 13818-1 is applied here as well.

Now that a dynamic payload type is to be used, RFC2327 [8], Session Decription
Protocol (SDP), should be used to broadcast the session information. More
specifically, the attribute parameter, a, in the SDP should be defined as 
follow:

       a=rtpmap:96 MP2T/27000000

Other parameters should be set followed the direction of RFC2327.


3. A four layer model:

When delivering MPEG transport stream using the profile defined in this draft, 
the buffer model given in Fig. 1 is assumed.

+-----------+     +-----------+	    +-----------+     +-----------+
|           |     |   MPEG    |	    |MPEG       |     |   MPEG    |
|RTP packet |---->| transport |---->|    PES    |---->|A/V Compr. |
|           |     |  packet   |	    |  packet   |     |   layer   |
+-----------+     +-----------+	    +-----------+     +-----------+


RTP timestamps     MPEG PCRs
      |                 |
      v	                v
+-----------+     +-----------+	    +-----------+     +-----------+
|    IP	    |     |   MPEG    | Vid.|   MPEG    |     |   MPEG	  |
| Dejitter  |---->| Transport |---->|Demultiplex|---->|Elementary |
|  Buffer   |     |  Buffer   |\    |  Buffer   |     |   Buffer  |
+-----------+     +-----------+ \   +-----------+     +-----------+
                        |        \                  
                        |         \ Audio
                        |          \
                        |           \                   +-----------+
                        |            \                  |   MPEG    |
                        |             -------------->   | Audio Main|
                        |                               |   Buffer  |
                        |                               +-----------+
                        |System info & other data
                        |                               +-----------+
                        |      	                        |   MPEG    |
                        +--------------------------->   |System Main|
                                                        |  Buffer   |
                                                        +-----------+

        Figure 1: The Packet/Buffer Model for MPEG MPTS over RTP


The Buffer model shown in Figure 1 is the extension of the buffer model given 
in ISO/IEC 13818-1. In addition to different MPEG packet layers, RTP layer and 
its associated IP dejitter buffer is added. The model given in Fig. 1 implies 
two clock recovery functions. The timestamps from RTP packets are used to 
synchronize the receiver clock for RTP session, while MPEG PCRs from MPEG 
transport packets are used to recover the MPEG decoder clock.

The size of the dejitter buffer is determined by the maximal delay variation 
introduced by the IP network. However, there are many ways to determine size of 
the dejitter buffer statically or statistically. For example, the jitter can be 
computed from the "interarrival jitter" from the sender report in its 
corresponding RTCP session. Alternatively, the jitter can also be computed from 
the guaranteed service block in ADSpec in the RSVP PATH message if RSVP is used 
for QoS. This document does not specify any method to determine the buffer size.

There are two benefits by using layered approach given in Fig. 1. The first 
benefit is that all the standardized tables and new services defined by DVB and 
ATSC can be included without any extra work. Thus, any video content that is 
available from digital broadcast industry can be easily delivered to the end 
user through IP network with minimal work at the RTP layer. The second benefit 
of this layered model is that the IP dejittering layer does not need to 
implement at the decoder side. For example, one can remove jitter induced by IP 
network first and broadcast MPEG MPTS to the end customer. The purpose of the 
IP dejitter buffer is to recover the temporal relationship between MPEG 
transport packet at the buffer output. By doing so, as long as the the MPEG 
MPTS is compliant at the ingress point, the output of the dejitter buffer 
should be compliant to ISO/IEC 13818-1. Thus, as long as QoS can be supported, 
the IP network can be viewed as another streaming transport method, just like 
ATM.


4. The Use of Timestamp

In addition to its original purpose of clock synchronization, when the payload 
type of 33 is used with marker bit set to one, the timestamp is also used for 
the following information:


a. Derived Piece-wise constant-bitrate(CBR) for the current RTP packet. It is 
assumed under this document that the session bitrate for any RTP packet is a 
constant. By knowing the size of RTP packet (from the length field in UDP 
packet), the timestamps from the current and the next RTP packet, this constant 
bitrate can be derived. However, the bitrate between packets can be different. 
Thus, a more constrained VBR can be implemented by changing the channel bitrate 
at the boundary of each RTP packet. 

b. Derived timestamp for every MPEG transport packets encapsulated in the RTP 
packet. Now that the session bitrate within one RTP packet is a constant and 
the timestamp of the RTP packet represents the timestamp of the first MPEG 
packet, it implies that the timestamp corresponding to the rest of MPEG packets 
in the RTP packet can be derived from it. As an example, suppose the current 
RTP packet has a timestamp of 200, the next RTP has a timestamp of 400, and the 
number of MPEG packets encapsulated in one RTP packet is five.  The first MPEG 
packet has a timestamp of 200, the second 240, the third 280, the fourth 320, 
and the fifth 360. Thus, the packet flowing out of the dejitter buffer is 
deterministic.

4. Limitation on the rate change when M is set to one

According to RFC2250, when M is set to one, it indicates that the session has
switched to different source. Thus, the receiver should reset its PLL and lock
to the new clock immediately.  The implication of this marker bit is that there
is no relationship between the current RTP packet and the previous RTP packet.
Thus, the rate change must be prohibited on the last RTP packet before
performing any source switching.

5. References

[1] D. Hoffman et al., "RTP Payload Format for MPEG1/MPEG2 Video", RFC 2250, 
    Jan. 1998.

[2] W. Fenner, "Internet Group Management Protocol Ver. 2", RFC2236, Nov. 
    1997.

[3] R. Braden et al., "Resource ReSerVation Protocol (RSVP) - Version 1 
    Functional Specification", RFC 2205, Sept. 1997.

[4] H. Schulzrinne, "RTP Profile for Audio and Video Conferences with Minimal 
    Control", RFC 1890, Jan. 1996.

[5] ISO/SEC. Generic Coding of Motion Pictures and Associated Audio: Systems. 
    ISO/IEC Standard 13818-1, 1995.

[6] ISO/IEC. Generic Coding of Motion Pictures and Associated Audio: Extension 
    for DSM-CC. ISO/IEC Standard 13818-6, 1998.

[7] CableLab. Data-Over-Cable Service Interface Specifications: Radio 
    Frequency Interface Specification. SP-RFIv1.1-I01-990311, 1999.

[8] M. Handley and V. Jacobson, "SDP: Session Description Protocol", RFC2327, 
    Apr. 1998.
--=====================_373887712==_
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From rem-conf-request@es.net  Fri Oct 29 03:00:39 1999
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Date: Thu, 28 Oct 1999 23:46:41 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: rem-conf@es.net
Subject: Re: Working group last call on RTP to Draft Standard
In-Reply-To: <Pine.WNT.4.10.9910272353280.-281261@revelstoke.cisco.com>
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To the AVT Working Group:

I should add a followup to my working group last call for comments on
the RTP spec:

*** If you have in the past sent comments to me or the other authors
*** of the RTP spec or profile which you feel have not been
*** appropriately addressed, then PLEASE repeat them now to the
*** rem-conf list, or to me individually if you prefer, because it is
*** entirely possible that I may have overlooked something.

Of course, I won't reopen issues that we have decided, but I want to
make sure there aren't corrections or clarifications that I have
missed.

As I mentioned, the changes to the RTP spec drafts were those we
discussed at the last meeting and/or on the list.  The changes made in
draft-ietf-avt-rtp-new-05 are:

  - The pseudo-code for the collision/loop detection algorithm in
    Section 8 has been changed to pseudo-C syntax to make the if-
    then-else structure more clear.

  - As discussed at the last meeting and on the list, text was added
    about keeping packets from the new address after a SSRC collision
    of two remote sources occurs.  This is for the case of mobile
    sources that change their IP address.

  - Reverse reconsideration is also used to possibly shorten the delay
    before sending RTCP SR when transitioning from passive receiver to
    active sender mode.

  - Timing out a participant is to be based on inactivity for a
    number of RTCP report intervals calculated using the receiver RTCP
    bandwidth fraction even for active senders.

  - Added text to say that active senders use RR packets in addition
    to SR packets when they have more than 31 sources to report on.

The changes made in draft-ietf-avt-profile-new-07 are:

  - The Comfort Noise (CN) static payload type number was changed from
    19 back to 13 as it was in the -01 version from which the value
    was incorporated into other groups' specifications.

  - The RTP timestamp clock rate for G722 was changed from the actual
    sampling rate of 16000 Hz back to 8000 Hz as it was in RFC 1890.
    A note has been added to clarify this discrepancy between the
    actual sampling rate and the RTP timestamp clock rate.

  - A note was added about MPA always using RTP timestamp clock rate
    90000 regardless of the audio sampling rate.

The changes to draft-ietf-avt-rtp-mime-01.txt are:

  - The mapping between MIME parameters and fmtp was clarified.

  - A registration form for the RED payload format was added.

  - An optional parameter "type" was added to the registration form
    for MPV.
							-- Steve




From rem-conf-request@es.net  Fri Oct 29 03:01:27 1999
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From: SNB Marketing <chucky22254@go.com>
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Subject: Advertise your product or website to millions free 
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Thanks,
SNB Marketing




From rem-conf-request@es.net  Fri Oct 29 03:07:06 1999
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From: Stephen Casner <casner@cisco.com>
To: Parekh Rishabh-crp049 <R.Parekh@motorola.com>
cc: "'rem-conf@es.net'" <rem-conf@es.net>,
        "Mike Korus (E-mail)" <cmk013@email.mot.com>
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On Wed, 27 Oct 1999, Parekh Rishabh-crp049 wrote:

> I have a question regarding the UDP ports used for RTP. Here is an excerpt
> from RFC 1889.
> 
> "RTP relies on the underlying protocol(s) to provide demultiplexing of RTP
> data and RTCP control streams. For UDP and similar protocols, RTP uses an
> even port number and the corresponding RTCP stream uses the next higher
> (odd) port number. If an application is supplied with an
> odd number for use as the RTP port, it should replace this number with the
> next lower (even) number."
> 
> Does RTP use an even UDP source port or an even UDP destination port? 

This paragraph refers to the destination port number.  I will clarify
this in the text.
							-- Steve




From rem-conf-request@es.net  Fri Oct 29 03:17:11 1999
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Date: Fri, 29 Oct 1999 00:07:09 -0700 (Pacific Daylight Time)
From: Stephen Casner <casner@cisco.com>
To: Murthy Nanduri <nvnmurthy@seeyes.co.in>
cc: rem-conf@es.net
Subject: Re: Padding in RTP
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On Wed, 27 Oct 1999, Murthy Nanduri wrote:

> I am working on the implementation of RFC2508.i.e., the
> RTP/UDP/IP Header compression and decompression.
...
> As per the RFC2508, I am setting the Bitmask for the Marker bit while
> sending the compressed RTP packet.

Right.

> When the padding bit is set, I am sending the compressed UDP packet..

If the padding bit is set on all the packets, then you should be able
to still send them as COMPRESSED_RTP packets.  You only need to send
COMPRESSED_UDP when the P bit changes (since there is no way to
indicate that in a COMPRESSED_RTP packet).

> Can you please let me know what has to be done by CRTP
> with the padding bytes?

Nothing.  They should be passed through just as they are, along with
the rest of the payload.

> I want to know whether the padding bytes have to be removed
> by CRTP compressor/decompressor OR it is the responsibility
> of the RTP applications to ignore the padding bytes..

The answers are no and yes.  A properly operating RFC2508 compressor
and decompressor will produce packets at the output of the
decompressor which are identical to the packets as they went into the
compressor.

> can U please let me know what for the Marker bit and Padding bits are
> used?

This is explained in the RTP spec (RFC 1889 or draft-ietf-avt-rtp-new-07)

							-- Steve




From rem-conf-request@es.net  Fri Oct 29 03:38:14 1999
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Date: Fri, 29 Oct 1999 12:12:21 +0530
From: Rafiq Shaikh <rafiq@chiplogic.com>
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Hi,

In the section 12 of RFC 1889, it says to refer to 'companion documents'
1. draft-ietf-avt-profile
2. RTP Payload formats for xyz audio/video encodings.

can anyone help me finding these docs? 
I am concerned about speech over RTP.

Thanks in advance.

Regards,
-Rafiq.



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Date: Thu, 28 Oct 1999 13:31:02 -0500
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: RTP port range question
In-reply-to: <5D7E7DD6AAEDD21187950008C791361DCB5F41@il02exm05.comm.mot.com>
To: Parekh Rishabh-crp049 <R.Parekh@motorola.com>, rem-conf@es.net
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RTP traditonally uses an even destination port number. I believe the source
port number to be indeterminate, and have seen even, odd, and downright
random implementations.

One of the big problems we're seeing in SIP-PSTN gateway work is port number
and the description thereof in SDP. As used, the SDP only specifies the
destination port number (and IP address) and says nothing about the source.
It is quite legal to use a model where, for one "end" of the conversation,
one host is listening and an entirely different host is sending. The other
end has no information about where the stream is coming from. This makes
writing a dyamically stateful firewall very difficult. It also caused a
number of SIP implementations to blow chunks at the last bakeoff.


--
Dean

> -----Original Message-----
> From: Parekh Rishabh-crp049 [mailto:R.Parekh@motorola.com]
> Sent: Wednesday, October 27, 1999 9:16 AM
> To: 'rem-conf@es.net'
> Cc: Mike Korus (E-mail)
> Subject: RTP port range question
>
>
> I have a question regarding the UDP ports used for RTP. Here is an excerpt
> from RFC 1889.
>
> "RTP relies on the underlying protocol(s) to provide demultiplexing of RTP
> data and RTCP control streams. For UDP and similar protocols, RTP uses an
> even port number and the corresponding RTCP stream uses the next higher
> (odd) port number. If an application is supplied with an
> odd number for use as the RTP port, it should replace this number with the
> next lower (even) number."
>
> Does RTP use an even UDP source port or an even UDP destination port?
>
> Thanks to all who reply,
> Rishabh Parekh
> Motorola
>




From rem-conf-request@es.net  Fri Oct 29 05:49:30 1999
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To: Rafiq Shaikh <rafiq@chiplogic.com>
cc: "rem-conf@es.net" <rem-conf@es.net>
Subject: Re: RTP Profiles and Payload formats
In-reply-to: Your message of "Fri, 29 Oct 1999 12:12:21 +0530." <381941CD.D64BCB9E@chiplogic.com>
Date: Fri, 29 Oct 1999 10:39:59 +0100
Message-ID: <1044.941189999@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Rafiq Shaikh writes:
>Hi,
>
>In the section 12 of RFC 1889, it says to refer to 'companion documents'
>1. draft-ietf-avt-profile

This was published as RFC1890, and is being updated as
http://www.ietf.org/internet-drafts/draft-ietf-avt-profile-new-07.txt

>2. RTP Payload formats for xyz audio/video encodings.

These are referenced from the audio/video profile, or you can find them at
http://www.ietf.org/html.charters/avt-charter.html

Colin



From rem-conf-request@es.net  Fri Oct 29 07:34:51 1999
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Date: Fri, 29 Oct 1999 13:09:46 +0200
From: Long Le <le@fokus.gmd.de>
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Henning Schulzrinne wrote:

> I'm looking for voice-activity detection software modules. I'm aware of
> the modules in vat, nevot and rat. Anything else?
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

My Diploma thesis under supervision of Henning Sanneck at GMD Fokus was
about
Internet speech transmission and also addresses the problem of
voiced/unvoiced and
silence detection (page 32-36).

ftp://ftp.fokus.gmd.de/pub/glone/papers/Le9906_Loss-resilient.pdf.gz

i only have an implementation in Matlab but implementation in C is really
straight forward.

-- long




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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload for Text Conversation
	Author(s)	: G. Hellstrom
	Filename	: draft-ietf-avt-rtp-text-02.txt
	Pages		: 9
	Date		: 28-Oct-99
	
This memo describes how to carry text conversation session contents
in RTP packets. Text conversation session contents is specified in
ITU-T Recommendation T.140 [1]. 
Text conversation is used alone or in connection to other
conversational facilities such as video and voice, to form multimedia
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This RTP payload description contains an optional possibility to
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From rem-conf-request@es.net  Fri Oct 29 10:42:56 1999
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Rafiq Shaikh wrote:
> 
> Hi,
> 
> In the section 12 of RFC 1889, it says to refer to 'companion documents'
> 1. draft-ietf-avt-profile
> 2. RTP Payload formats for xyz audio/video encodings.
> 
> can anyone help me finding these docs?
> I am concerned about speech over RTP.

Related information and a reasonably complete listing of Internet Drafts
can be found on the RTP page at http://www.cs.columbia.edu/~hgs/rtp

Current drafts are also available through the IETF web site, but not all
RTP-related drafts are listed on the AVT working group web page.
(Individual submissions are not listed there.)

> 
> Thanks in advance.
> 
> Regards,
> -Rafiq.



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YOU COULD REGISTER ON-LINE VIA www.lrg.ufsc.br/~lanoms99.

***** IEEE LANOMS99 - TUTORIALS *****

TUTORIAL 1.1 - The technological and Economic Effects of the Deregulation
               of the Global Telecommunications Industry upon the Application
               of the Principles of Network Management
By Keith von Mecklenburg (St. Andrews University, Scotland)     
Auditorium#1 - Dec 05 - 09:00 - 12:30

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***** IEEE LANOMS99 - TUTORIALS ABSTRACS *****

TUTORIAL 1.1 - The Technological and Economic Effects of the Deregulation 
of the Global Telecommunications Industry upon the Application of the
Principles of Network Management

Deregulation has increased the level of competition within National and
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>From this tutorial you will learn network management models, neural
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TUTORIAL 2.1 - New Global Solutions of Event Correlation based on
Distributed  infrastructure 

Audience: This is an introductory/intermediate level tutorial. The intended
audience includes network/service managers and technical operations support
professionals interested in learning new advanced methods of fault, 
performance.
Abstract: Event correlation is a widely accepted technology for managing the
complexity of modern telecommunication and data networks. Specific 
correlation applications have been developed to manage switched, SS7, 
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network management arena have either developed their own, usually 
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products such as InCharge, NerveCenter, ILOG, ART-Enterprise, RTWorks, 
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operations support functions, based on real-time event correlation have 
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correlation systems, there is a need to re-evaluate the current event 
correlation solutions and define new directions. It is important to 
broaden the utility of event correlation to other aspects of network 
management, such as performance, configuration, testing, security, and 
service quality management. Most importantly, the technology should be 
extended to perform cross-correlation between different domains, e.g. 
cross-correlation This tutorial is divided into three parts. In the first 
introductory part we will review the tasks, objectives, major 
technologies and products of existing solutions of event correlation. In 
the second part, we will introduce the concept, architectural
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of event correlation  ith practical network management examples. 

TUTORIAL 2.2 - Network Management with SNMP: The Manager-Agent and
Distributed Paradigms 

[Abstract]
This tutorial starts with an introduction to the SNMP network management
framework, describing its main components, including the SMI, MIB's,
agents, management applications and the protocol itself.  Following this
introduction, the distributed paradigm for network management is examined,
as well as different approaches for its deployment using SNMP. Case
studies are presented, including the Routing Proxy, the Delegado MIB
and the implementation of distributed system-level diagnosis using
SNMP. Finally, an overview of the IETF's DisMan MIB's is presented.
[Audience]
This is an introdutory/intermediate tutorial.  The intended audience
includes network managers and technical professionals interested in
learning about SNMP and how it can be used both a manager-agent and a
distributed management paradigm.   

***** IEEE LANOMS99 - TUTORIALS REGISTRATION FORM ****

NAME:
TITLE:
ORGANIZATION:
DEPARTMENT:
ADDRESS:
STATE:
COUNTRY:
EMAIL:
TELEPHONE:
FAX:       

Number of tutorial to be Registrated
IEEE Members     ( ) USD 125 - 1 Tutorial or ( ) USD 225 - 2 Tutorials
IEEE Non-Members ( ) USD 150 - 1 Tutorial or ( ) USD 250 - 2 Tutorials
Tutorials (Mark your choices)
( ) Tutorial 1.1 or ( ) Tutorial 1.2
( ) Tutorial 2.1 or ( ) Tutorial 2.1

Total Tutorial: USD________

Method of Payment
( ) Method 1 - Money order in USD Payable to FEESC
( ) Method 2 - Check Brazilian Banks in R$ (exchange rate 1,9 R$/USD)
For both methods make transfer to the following bank account:  
Bank Name: Banco do Brasil
Bank Number: 001
Branch Number: 1453-2
Bank Account: 204.422.6
Swift Code: BRASBRRJFNS
( ) Method 3 - ( ) Mastercard
Number:__________________
Expiration Date:______________
Account Name:________________
Authorized ( ) I authorize FEESC to deduct the amount inserted at
TOTAL REMITTANCE from my credit card.

SEND BY FAX OR BY EMAIL THE COMPLETED FORM WITH PAYMENT TO:
Fundacao de Ensino e Engenharia de Santa Catarina (FEESC)
Caixa Postal 5040
88040-970, Florianopolis (SC) Brazil
Phone: +55.48.2334455
FAX: +55.48.3319677
EMAIL: katia@feesc.ufsc.br




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Date: Sat, 30 Oct 1999 15:48:21 -0200 (EDT)
From: "LANOMS'99" <lanoms99@lrg.ufsc.br>
Subject: IEEE LANOMS99 - Tutorials Info and Registration Form
To: Willie LU <wwlu@ieee.org>
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YOU COULD REGISTER ON-LINE VIA www.lrg.ufsc.br/~lanoms99.

***** IEEE LANOMS99 - TUTORIALS *****

TUTORIAL 1.1 - The technological and Economic Effects of the Deregulation
               of the Global Telecommunications Industry upon the Application
               of the Principles of Network Management
By Keith von Mecklenburg (St. Andrews University, Scotland)     
Auditorium#1 - Dec 05 - 09:00 - 12:30

TUTORIAL 1.2 - Neural Networks and Distributed Algorithms for Network
               Management
By Jayantha Herath (St. Cloud State University, USA), and
By Susantha Herath (Aizu University, Japan) 
Auditorium#2 - Dec 05 - 09:00 - 12:30

TUTORIAL 2.1 - New Global Solutions of Event Correlation based on
               Distributed  infrastructure
By Gabriel Jakobson (GTE, USA)
Auditorium#1 - Dec 05 - 14:00 - 17:30

TUTORIAL 2.2 - Network Management with SNMP: The Manager-Agent and
               Distributed Paradigms
By Elias P. Duarte Jr. (Federal University of Parana, Brazil)
Auditorium#2 - Dec 05 - 14:00 - 17:30


***** IEEE LANOMS99 - TUTORIALS ABSTRACS *****

TUTORIAL 1.1 - The Technological and Economic Effects of the Deregulation 
of the Global Telecommunications Industry upon the Application of the
Principles of Network Management

Deregulation has increased the level of competition within National and
International Telecommunications Industries, as State firms (e.g. Deutsche
Telekom) and those private firms that dominate the telecoms market ( e.g.
the Bell Companies and BT) within national jurisdictions, have to consider
the lowering of line access charges to meet that competitive situation.
It means that all firms, incumbents and new entrants, have to be focussed
on the level of fixed and variable costs. Instead of being able to design and
implement technological network systems with little regard to costs, the
pressure is upon management to find the most effective technological
solution which will also meet the measure of the lowest fixed and variable
costs.   
The tutorial will therefore advance a new set of network management 
principles to meet the new and developing market competitive situation. 
Network Management will seek to develop operational models which satisfy the
conflicting demands of the best technological and the best economic models
of effective Network Management.


TUTORIAL 1.2 - Neural Networks and Distributed algorithms for Network 
Management

This tutorial provides an overview to network management issues in high
performance distributed computing which promises wide range of new
computationally intensive applications such as intelligent agents, data
mining, computing at remote sites across wide variety of networks
including the web. Network management and development of such applications
and tools are neither easy nor quick because of the heterogeneous nature
of the distributed systems. The problem becomes much difficult because of
the service needs of distributed computations to be considered in the
design and implementation process.  Such needs include locating resources,
allocating resources, communication protocols, security, accessibility,
files, faults, memories and control. Network management is crucial to
achieving highest performance from a dynamically changing distributed 
computing environment.
This tutorial addresses the importance of applying learning techniques to
distributed algorithms for network management. After providing an overview
of centralized, hierarchical and distributed management models, protocols
and architectures it will introduce the importance of integrating learning
algorithms to the distributed algorithms. It will describe neural network
based computations as one out of many learning techniques available for
such applications. The focus will be to examine the integration of
learning techniques to distributed algorithms for network management. This
tutorial will also overview the recent activities and issues in the areas
of communication protocols, synchronization algorithms for clocks,
elections and deadlock handling, scheduling and fault tolerance of
distributed processes and processors, sharing and replication management
of distributed file systems and distributed shared memories.
>From this tutorial you will learn network management models, neural
network algorithms and  distributed algorithms for network management.


TUTORIAL 2.1 - New Global Solutions of Event Correlation based on
Distributed  infrastructure 

Audience: This is an introductory/intermediate level tutorial. The intended
audience includes network/service managers and technical operations support
professionals interested in learning new advanced methods of fault, 
performance.
Abstract: Event correlation is a widely accepted technology for managing the
complexity of modern telecommunication and data networks. Specific 
correlation applications have been developed to manage switched, SS7, 
wireless, ATM, SONET, IP, and other networks. All major players in the 
network management arena have either developed their own, usually 
embedded event correlation procedures, or have used event correlation 
products such as InCharge, NerveCenter, ILOG, ART-Enterprise, RTWorks, 
NetExpert, and G2. Various approaches to event correlation exist, 
including rule-, case-, and model-based reasoning, finite Major network 
operations support functions, based on real-time event correlation have 
emerged, including reduction of event flow by context-sensitive event 
filtering, fault management by fault diagnostics, localization, root 
cause identification and generation of corrective actions, and status
reporting of complex multi-level hierarchical networks by semantic 
generalization of Despite the progress in the development of event 
correlation systems, there is a need to re-evaluate the current event 
correlation solutions and define new directions. It is important to 
broaden the utility of event correlation to other aspects of network 
management, such as performance, configuration, testing, security, and 
service quality management. Most importantly, the technology should be 
extended to perform cross-correlation between different domains, e.g. 
cross-correlation This tutorial is divided into three parts. In the first 
introductory part we will review the tasks, objectives, major 
technologies and products of existing solutions of event correlation. In 
the second part, we will introduce the concept, architectural
framework, and components of a distributed global event correlation 
solution. In the third part we will illustrate how the concepts and 
architectural framework were implemented in a particular system GRACE, 
developed at GTE Laboratories. We will discuss the use of CORBA, Java, 
XML, SWING, Jacc and other technologies in GRACE, and illustrate the use 
of event correlation  ith practical network management examples. 

TUTORIAL 2.2 - Network Management with SNMP: The Manager-Agent and
Distributed Paradigms 

[Abstract]
This tutorial starts with an introduction to the SNMP network management
framework, describing its main components, including the SMI, MIB's,
agents, management applications and the protocol itself.  Following this
introduction, the distributed paradigm for network management is examined,
as well as different approaches for its deployment using SNMP. Case
studies are presented, including the Routing Proxy, the Delegado MIB
and the implementation of distributed system-level diagnosis using
SNMP. Finally, an overview of the IETF's DisMan MIB's is presented.
[Audience]
This is an introdutory/intermediate tutorial.  The intended audience
includes network managers and technical professionals interested in
learning about SNMP and how it can be used both a manager-agent and a
distributed management paradigm.   

***** IEEE LANOMS99 - TUTORIALS REGISTRATION FORM ****

NAME:
TITLE:
ORGANIZATION:
DEPARTMENT:
ADDRESS:
STATE:
COUNTRY:
EMAIL:
TELEPHONE:
FAX:       

Number of tutorial to be Registrated
IEEE Members     ( ) USD 125 - 1 Tutorial or ( ) USD 225 - 2 Tutorials
IEEE Non-Members ( ) USD 150 - 1 Tutorial or ( ) USD 250 - 2 Tutorials
Tutorials (Mark your choices)
( ) Tutorial 1.1 or ( ) Tutorial 1.2
( ) Tutorial 2.1 or ( ) Tutorial 2.1

Total Tutorial: USD________

Method of Payment
( ) Method 1 - Money order in USD Payable to FEESC
( ) Method 2 - Check Brazilian Banks in R$ (exchange rate 1,9 R$/USD)
For both methods make transfer to the following bank account:  
Bank Name: Banco do Brasil
Bank Number: 001
Branch Number: 1453-2
Bank Account: 204.422.6
Swift Code: BRASBRRJFNS
( ) Method 3 - ( ) Mastercard
Number:__________________
Expiration Date:______________
Account Name:________________
Authorized ( ) I authorize FEESC to deduct the amount inserted at
TOTAL REMITTANCE from my credit card.

SEND BY FAX OR BY EMAIL THE COMPLETED FORM WITH PAYMENT TO:
Fundacao de Ensino e Engenharia de Santa Catarina (FEESC)
Caixa Postal 5040
88040-970, Florianopolis (SC) Brazil
Phone: +55.48.2334455
FAX: +55.48.3319677
EMAIL: katia@feesc.ufsc.br




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From: Publishers-Company@excite.com
Message-Id: <199910310930.RAA00149@kedah.gov.my>
To: PublishingCompany1@excite.com
Date: Sun, 31 Oct 99 01:52:57 EST
Subject: Publishing Company for Sale!
Reply-To: removeforever1@bigfoot.com
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